public int write(byte[] audioData, int offsetInBytes, int sizeInBytes) {
if (mState == STATE_UNINITIALIZED) {
return ERROR_INVALID_OPERATION;
}
if ( (audioData == null) || (offsetInBytes < 0 ) || (sizeInBytes < 0)
|| (offsetInBytes + sizeInBytes < 0) // detect integer overflow
|| (offsetInBytes + sizeInBytes > audioData.length)) {
return ERROR_BAD_VALUE;
}
int ret = native_write_byte(audioData, offsetInBytes, sizeInBytes, mAudioFormat);
if ((mDataLoadMode == MODE_STATIC)
&& (mState == STATE_NO_STATIC_DATA)
&& (ret > 0)) {
// benign race with respect to other APIs that read mState
mState = STATE_INITIALIZED;
}
return ret;
}
static jint android_media_AudioTrack_native_write_byte(JNIEnv *env, jobject thiz,
jbyteArray javaAudioData,
jint offsetInBytes, jint sizeInBytes,
jint javaAudioFormat) {
//ALOGV("android_media_AudioTrack_native_write_byte(offset=%d, sizeInBytes=%d) called",
// offsetInBytes, sizeInBytes);
sp<AudioTrack> lpTrack = getAudioTrack(env, thiz);
if (lpTrack == NULL) {
jniThrowException(env, "java/lang/IllegalStateException",
"Unable to retrieve AudioTrack pointer for write()");
return 0;
}
// get the pointer for the audio data from the java array
// NOTE: We may use GetPrimitiveArrayCritical() when the JNI implementation changes in such
// a way that it becomes much more efficient. When doing so, we will have to prevent the
// AudioSystem callback to be called while in critical section (in case of media server
// process crash for instance)
jbyte* cAudioData = NULL;
if (javaAudioData) {
cAudioData = (jbyte *)env->GetByteArrayElements(javaAudioData, NULL);
if (cAudioData == NULL) {
ALOGE("Error retrieving source of audio data to play, can't play");
return 0; // out of memory or no data to load
}
} else {
ALOGE("NULL java array of audio data to play, can't play");
return 0;
}
jint written = writeToTrack(lpTrack, javaAudioFormat, cAudioData, offsetInBytes, sizeInBytes);
env->ReleaseByteArrayElements(javaAudioData, cAudioData, 0);
//ALOGV("write wrote %d (tried %d) bytes in the native AudioTrack with offset %d",
// (int)written, (int)(sizeInBytes), (int)offsetInBytes);
return written;
}
jint writeToTrack(const sp<AudioTrack>& track, jint audioFormat, jbyte* data,
jint offsetInBytes, jint sizeInBytes) {
// give the data to the native AudioTrack object (the data starts at the offset)
ssize_t written = 0;
// regular write() or copy the data to the AudioTrack's shared memory?
if (track->sharedBuffer() == 0) {
written = track->write(data + offsetInBytes, sizeInBytes);
// for compatibility with earlier behavior of write(), return 0 in this case
if (written == (ssize_t) WOULD_BLOCK) {
written = 0;
}
} else {
if (audioFormat == ENCODING_PCM_16BIT) {
// writing to shared memory, check for capacity
if ((size_t)sizeInBytes > track->sharedBuffer()->size()) {
sizeInBytes = track->sharedBuffer()->size();
}
memcpy(track->sharedBuffer()->pointer(), data + offsetInBytes, sizeInBytes);
written = sizeInBytes;
} else if (audioFormat == ENCODING_PCM_8BIT) {
// data contains 8bit data we need to expand to 16bit before copying
// to the shared memory
// writing to shared memory, check for capacity,
// note that input data will occupy 2X the input space due to 8 to 16bit conversion
if (((size_t)sizeInBytes)*2 > track->sharedBuffer()->size()) {
sizeInBytes = track->sharedBuffer()->size() / 2;
}
int count = sizeInBytes;
int16_t *dst = (int16_t *)track->sharedBuffer()->pointer();
const int8_t *src = (const int8_t *)(data + offsetInBytes);
while (count--) {
*dst++ = (int16_t)(*src++^0x80) << 8;
}
// even though we wrote 2*sizeInBytes, we only report sizeInBytes as written to hide
// the 8bit mixer restriction from the user of this function
written = sizeInBytes;
}
}
return written;
}
ssize_t AudioTrack::write(const void* buffer, size_t userSize)
{
if (mTransfer != TRANSFER_SYNC || mIsTimed) {
return INVALID_OPERATION;
}
if (ssize_t(userSize) < 0 || (buffer == NULL && userSize != 0)) {
// Sanity-check: user is most-likely passing an error code, and it would
// make the return value ambiguous (actualSize vs error).
ALOGE("AudioTrack::write(buffer=%p, size=%zu (%zd)", buffer, userSize, userSize);
return BAD_VALUE;
}
size_t written = 0;
Buffer audioBuffer;
while (userSize >= mFrameSize) {
audioBuffer.frameCount = userSize / mFrameSize;
//获得Buffer
status_t err = obtainBuffer(&audioBuffer, &ClientProxy::kForever);
if (err < 0) {
if (written > 0) {
break;
}
return ssize_t(err);
}
size_t toWrite;
if (mFormat == AUDIO_FORMAT_PCM_8_BIT && !(mFlags & AUDIO_OUTPUT_FLAG_DIRECT)) {
// Divide capacity by 2 to take expansion into account
toWrite = audioBuffer.size >> 1;
memcpy_to_i16_from_u8(audioBuffer.i16, (const uint8_t *) buffer, toWrite);
} else {
toWrite = audioBuffer.size;
memcpy(audioBuffer.i8, buffer, toWrite);
}
buffer = ((const char *) buffer) + toWrite;
userSize -= toWrite;
written += toWrite;
releaseBuffer(&audioBuffer);
}
return written;
}
status_t obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
struct timespec *elapsed = NULL, size_t *nonContig = NULL);
status_t AudioTrack::obtainBuffer(Buffer* audioBuffer, const struct timespec *requested,
struct timespec *elapsed, size_t *nonContig)
{
// previous and new IAudioTrack sequence numbers are used to detect track re-creation
uint32_t oldSequence = 0;
uint32_t newSequence;
Proxy::Buffer buffer;
status_t status = NO_ERROR;
static const int32_t kMaxTries = 5;
int32_t tryCounter = kMaxTries;
do {
// obtainBuffer() is called with mutex unlocked, so keep extra references to these fields to
// keep them from going away if another thread re-creates the track during obtainBuffer()
sp<AudioTrackClientProxy> proxy;
sp<IMemory> iMem;
{ // start of lock scope
AutoMutex lock(mLock);
newSequence = mSequence;
// did previous obtainBuffer() fail due to media server death or voluntary invalidation?
if (status == DEAD_OBJECT) {
// re-create track, unless someone else has already done so
if (newSequence == oldSequence) {
status = restoreTrack_l("obtainBuffer");
if (status != NO_ERROR) {
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
}
}
oldSequence = newSequence;
// Keep the extra references
proxy = mProxy;
iMem = mCblkMemory;
if (mState == STATE_STOPPING) {
status = -EINTR;
buffer.mFrameCount = 0;
buffer.mRaw = NULL;
buffer.mNonContig = 0;
break;
}
// Non-blocking if track is stopped or paused
if (mState != STATE_ACTIVE) {
requested = &ClientProxy::kNonBlocking;
}
} // end of lock scope
buffer.mFrameCount = audioBuffer->frameCount;
// FIXME starts the requested timeout and elapsed over from scratch
status = proxy->obtainBuffer(&buffer, requested, elapsed);
} while ((status == DEAD_OBJECT) && (tryCounter-- > 0));
audioBuffer->frameCount = buffer.mFrameCount;
audioBuffer->size = buffer.mFrameCount * mFrameSizeAF;
audioBuffer->raw = buffer.mRaw;
if (nonContig != NULL) {
*nonContig = buffer.mNonContig;
}
return status;
}
status_t AudioTrack::restoreTrack_l(const char *from)
{
ALOGW("dead IAudioTrack, %s, creating a new one from %s()",
isOffloaded() ? "Offloaded" : "PCM", from);
++mSequence;
status_t result;
// refresh the audio configuration cache in this process to make sure we get new
// output parameters in getOutput_l() and createTrack_l()
AudioSystem::clearAudioConfigCache();
if (isOffloaded()) {
return DEAD_OBJECT;
}
// force new output query from audio policy manager;
mOutput = 0;
audio_io_handle_t output = getOutput_l();
// if the new IAudioTrack is created, createTrack_l() will modify the
// following member variables: mAudioTrack, mCblkMemory and mCblk.
// It will also delete the strong references on previous IAudioTrack and IMemory
// take the frames that will be lost by track recreation into account in saved position
size_t position = mProxy->getPosition() + mProxy->getFramesFilled();
size_t bufferPosition = mStaticProxy != NULL ? mStaticProxy->getBufferPosition() : 0;
result = createTrack_l(mStreamType,
mSampleRate,
mFormat,
mReqFrameCount, // so that frame count never goes down
mFlags,
mSharedBuffer,
output,
position /*epoch*/);
if (result == NO_ERROR) {
// continue playback from last known position, but
// don't attempt to restore loop after invalidation; it's difficult and not worthwhile
if (mStaticProxy != NULL) {
mLoopPeriod = 0;
mStaticProxy->setLoop(bufferPosition, mFrameCount, 0);
}
// FIXME How do we simulate the fact that all frames present in the buffer at the time of
// track destruction have been played? This is critical for SoundPool implementation
// This must be broken, and needs to be tested/debugged.
if (mState == STATE_ACTIVE) {
result = mAudioTrack->start();
}
}
if (result != NO_ERROR) {
//Use of direct and offloaded output streams is ref counted by audio policy manager.
// As getOutput was called above and resulted in an output stream to be opened,
// we need to release it.
AudioSystem::releaseOutput(output);
ALOGW("restoreTrack_l() failed status %d", result);
mState = STATE_STOPPED;
}
return result;
}
audio_io_handle_t AudioTrack::getOutput_l()
{
if (mOutput) {
return mOutput;
} else {
return AudioSystem::getOutput(mStreamType,
mSampleRate, mFormat, mChannelMask, mFlags);
}
}
size_t getPosition() {
return mEpoch + mCblk->mServer;
}
size_t ClientProxy::getFramesFilled() {
audio_track_cblk_t* cblk = mCblk;
int32_t front;
int32_t rear;
if (mIsOut) {
front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
rear = cblk->u.mStreaming.mRear;
} else {
rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
front = cblk->u.mStreaming.mFront;
}
ssize_t filled = rear - front;
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
return 0;
}
return (size_t)filled;
}
size_t StaticAudioTrackClientProxy::getBufferPosition()
{
size_t bufferPosition;
if (mMutator.ack()) {
bufferPosition = (size_t) mCblk->u.mStatic.mBufferPosition;
if (bufferPosition > mFrameCount) {
bufferPosition = mFrameCount;
}
} else {
bufferPosition = mBufferPosition;
}
return bufferPosition;
}
// To facilitate quicker recovery from server failure, this value limits the timeout per each futex
// wait. However it does not protect infinite timeouts. If defined to be zero, there is no limit.
// FIXME May not be compatible with audio tunneling requirements where timeout should be in the
// order of minutes.
#define MAX_SEC 5
status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *requested,
struct timespec *elapsed)
{
LOG_ALWAYS_FATAL_IF(buffer == NULL || buffer->mFrameCount == 0);
struct timespec total; // total elapsed time spent waiting
total.tv_sec = 0;
total.tv_nsec = 0;
bool measure = elapsed != NULL; // whether to measure total elapsed time spent waiting
status_t status;
enum {
TIMEOUT_ZERO, // requested == NULL || *requested == 0
TIMEOUT_INFINITE, // *requested == infinity
TIMEOUT_FINITE, // 0 < *requested < infinity
TIMEOUT_CONTINUE, // additional chances after TIMEOUT_FINITE
} timeout;
if (requested == NULL) {
timeout = TIMEOUT_ZERO;
} else if (requested->tv_sec == 0 && requested->tv_nsec == 0) {
timeout = TIMEOUT_ZERO;
} else if (requested->tv_sec == INT_MAX) {
timeout = TIMEOUT_INFINITE;
} else {
timeout = TIMEOUT_FINITE;
if (requested->tv_sec > 0 || requested->tv_nsec >= MEASURE_NS) {
measure = true;
}
}
struct timespec before;
bool beforeIsValid = false;
audio_track_cblk_t* cblk = mCblk;
bool ignoreInitialPendingInterrupt = true;
// check for shared memory corruption
if (mIsShutdown) {
status = NO_INIT;
goto end;
}
for (;;) {
int32_t flags = android_atomic_and(~CBLK_INTERRUPT, &cblk->mFlags);
// check for track invalidation by server, or server death detection
if (flags & CBLK_INVALID) {
ALOGV("Track invalidated");
status = DEAD_OBJECT;
goto end;
}
// check for obtainBuffer interrupted by client
if (!ignoreInitialPendingInterrupt && (flags & CBLK_INTERRUPT)) {
ALOGV("obtainBuffer() interrupted by client");
status = -EINTR;
goto end;
}
ignoreInitialPendingInterrupt = false;
// compute number of frames available to write (AudioTrack) or read (AudioRecord)
int32_t front;
int32_t rear;
if (mIsOut) {
// The barrier following the read of mFront is probably redundant.
// We're about to perform a conditional branch based on 'filled',
// which will force the processor to observe the read of mFront
// prior to allowing data writes starting at mRaw.
// However, the processor may support speculative execution,
// and be unable to undo speculative writes into shared memory.
// The barrier will prevent such speculative execution.
front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
rear = cblk->u.mStreaming.mRear;
} else {
// On the other hand, this barrier is required.
rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
front = cblk->u.mStreaming.mFront;
}
ssize_t filled = rear - front;
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
ALOGE("Shared memory control block is corrupt (filled=%d); shutting down", filled);
mIsShutdown = true;
status = NO_INIT;
goto end;
}
// don't allow filling pipe beyond the nominal size
size_t avail = mIsOut ? mFrameCount - filled : filled;
if (avail > 0) {
// 'avail' may be non-contiguous, so return only the first contiguous chunk
size_t part1;
if (mIsOut) {
rear &= mFrameCountP2 - 1;
part1 = mFrameCountP2 - rear;
} else {
front &= mFrameCountP2 - 1;
part1 = mFrameCountP2 - front;
}
if (part1 > avail) {
part1 = avail;
}
if (part1 > buffer->mFrameCount) {
part1 = buffer->mFrameCount;
}
buffer->mFrameCount = part1;
buffer->mRaw = part1 > 0 ?
&((char *) mBuffers)[(mIsOut ? rear : front) * mFrameSize] : NULL;
buffer->mNonContig = avail - part1;
mUnreleased = part1;
status = NO_ERROR;
break;
}
struct timespec remaining;
const struct timespec *ts;
switch (timeout) {
case TIMEOUT_ZERO:
status = WOULD_BLOCK;
goto end;
case TIMEOUT_INFINITE:
ts = NULL;
break;
case TIMEOUT_FINITE:
timeout = TIMEOUT_CONTINUE;
if (MAX_SEC == 0) {
ts = requested;
break;
}
// fall through
case TIMEOUT_CONTINUE:
// FIXME we do not retry if requested < 10ms? needs documentation on this state machine
if (!measure || requested->tv_sec < total.tv_sec ||
(requested->tv_sec == total.tv_sec && requested->tv_nsec <= total.tv_nsec)) {
status = TIMED_OUT;
goto end;
}
remaining.tv_sec = requested->tv_sec - total.tv_sec;
if ((remaining.tv_nsec = requested->tv_nsec - total.tv_nsec) < 0) {
remaining.tv_nsec += 1000000000;
remaining.tv_sec++;
}
if (0 < MAX_SEC && MAX_SEC < remaining.tv_sec) {
remaining.tv_sec = MAX_SEC;
remaining.tv_nsec = 0;
}
ts = &remaining;
break;
default:
LOG_FATAL("obtainBuffer() timeout=%d", timeout);
ts = NULL;
break;
}
int32_t old = android_atomic_and(~CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
if (measure && !beforeIsValid) {
clock_gettime(CLOCK_MONOTONIC, &before);
beforeIsValid = true;
}
errno = 0;
(void) syscall(__NR_futex, &cblk->mFutex,
mClientInServer ? FUTEX_WAIT_PRIVATE : FUTEX_WAIT, old & ~CBLK_FUTEX_WAKE, ts);
// update total elapsed time spent waiting
if (measure) {
struct timespec after;
clock_gettime(CLOCK_MONOTONIC, &after);
total.tv_sec += after.tv_sec - before.tv_sec;
long deltaNs = after.tv_nsec - before.tv_nsec;
if (deltaNs < 0) {
deltaNs += 1000000000;
total.tv_sec--;
}
if ((total.tv_nsec += deltaNs) >= 1000000000) {
total.tv_nsec -= 1000000000;
total.tv_sec++;
}
before = after;
beforeIsValid = true;
}
switch (errno) {
case 0: // normal wakeup by server, or by binderDied()
case EWOULDBLOCK: // benign race condition with server
case EINTR: // wait was interrupted by signal or other spurious wakeup
case ETIMEDOUT: // time-out expired
// FIXME these error/non-0 status are being dropped
break;
default:
status = errno;
ALOGE("%s unexpected error %s", __func__, strerror(status));
goto end;
}
}
}
end:
if (status != NO_ERROR) {
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
buffer->mNonContig = 0;
mUnreleased = 0;
}
if (elapsed != NULL) {
*elapsed = total;
}
if (requested == NULL) {
requested = &kNonBlocking;
}
if (measure) {
ALOGV("requested %ld.%03ld elapsed %ld.%03ld",
requested->tv_sec, requested->tv_nsec / 1000000,
total.tv_sec, total.tv_nsec / 1000000);
}
return status;
}
void AudioTrack::releaseBuffer(Buffer* audioBuffer)
{
if (mTransfer == TRANSFER_SHARED) {
return;
}
size_t stepCount = audioBuffer->size / mFrameSizeAF;
if (stepCount == 0) {
return;
}
Proxy::Buffer buffer;
buffer.mFrameCount = stepCount;
buffer.mRaw = audioBuffer->raw;
AutoMutex lock(mLock);
mInUnderrun = false;
mProxy->releaseBuffer(&buffer);
// restart track if it was disabled by audioflinger due to previous underrun
if (mState == STATE_ACTIVE) {
audio_track_cblk_t* cblk = mCblk;
if (android_atomic_and(~CBLK_DISABLED, &cblk->mFlags) & CBLK_DISABLED) {
ALOGW("releaseBuffer() track %p name=%s disabled due to previous underrun, restarting",
this, mName.string());
// FIXME ignoring status
mAudioTrack->start();
}
}
}
void ClientProxy::releaseBuffer(Buffer* buffer)
{
LOG_ALWAYS_FATAL_IF(buffer == NULL);
size_t stepCount = buffer->mFrameCount;
if (stepCount == 0 || mIsShutdown) {
// prevent accidental re-use of buffer
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
buffer->mNonContig = 0;
return;
}
LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased && mUnreleased <= mFrameCount));
mUnreleased -= stepCount;
audio_track_cblk_t* cblk = mCblk;
// Both of these barriers are required
if (mIsOut) {
//播放时，更新写入数据的指针，应该会同时通知数据读取端？
int32_t rear = cblk->u.mStreaming.mRear;
android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear);
} else {
//录音时，更新读取数据的指针，应该会同时通知数据写入端？
int32_t front = cblk->u.mStreaming.mFront;
android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront);
}
}