/*
* Copyright (C) 2007 The Android Open Source Project
*
* Licensed under the Apache License, Version 2.0 (the "License");
* you may not use this file except in compliance with the License.
* You may obtain a copy of the License at
*
* http://www.apache.org/licenses/LICENSE-2.0
*
* Unless required by applicable law or agreed to in writing, software
* distributed under the License is distributed on an "AS IS" BASIS,
* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
* See the License for the specific language governing permissions and
* limitations under the License.
*/
#define LOG_TAG "AudioTrackShared"
//#define LOG_NDEBUG 0
#include <private/media/AudioTrackShared.h>
#include <utils/Log.h>
#include <linux/futex.h>
#include <sys/syscall.h>
namespace android {
// used to clamp a value to size_t. TODO: move to another file.
template <typename T>
size_t clampToSize(T x) {
return sizeof(T) > sizeof(size_t) && x > (T) SIZE_MAX ? SIZE_MAX : x < 0 ? 0 : (size_t) x;
}
// incrementSequence is used to determine the next sequence value
// for the loop and position sequence counters. It should return
// a value between "other" + 1 and "other" + INT32_MAX, the choice of
// which needs to be the "least recently used" sequence value for "self".
// In general, this means (new_self) returned is max(self, other) + 1.
static uint32_t incrementSequence(uint32_t self, uint32_t other) {
int32_t diff = (int32_t) self - (int32_t) other;
if (diff >= 0 && diff < INT32_MAX) {
return self + 1; // we're already ahead of other.
}
return other + 1; // we're behind, so move just ahead of other.
}
audio_track_cblk_t::audio_track_cblk_t()
: mServer(0), mFutex(0), mMinimum(0)
, mVolumeLR(GAIN_MINIFLOAT_PACKED_UNITY), mSampleRate(0), mSendLevel(0)
, mBufferSizeInFrames(0)
, mFlags(0)
{
memset(&u, 0, sizeof(u));
}
// ---------------------------------------------------------------------------
Proxy::Proxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount, size_t frameSize,
bool isOut, bool clientInServer)
: mCblk(cblk), mBuffers(buffers), mFrameCount(frameCount), mFrameSize(frameSize),
mFrameCountP2(roundup(frameCount)), mIsOut(isOut), mClientInServer(clientInServer),
mIsShutdown(false), mUnreleased(0)
{
}
// ---------------------------------------------------------------------------
ClientProxy::ClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
size_t frameSize, bool isOut, bool clientInServer)
: Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer)
, mEpoch(0)
, mTimestampObserver(&cblk->mExtendedTimestampQueue)
{
setBufferSizeInFrames(frameCount);
}
const struct timespec ClientProxy::kForever = {INT_MAX /*tv_sec*/, 0 /*tv_nsec*/};
const struct timespec ClientProxy::kNonBlocking = {0 /*tv_sec*/, 0 /*tv_nsec*/};
#define MEASURE_NS 10000000 // attempt to provide accurate timeouts if requested >= MEASURE_NS
// 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
uint32_t ClientProxy::setBufferSizeInFrames(uint32_t size)
{
// The minimum should be greater than zero and less than the size
// at which underruns will occur.
const uint32_t minimum = 16; // based on AudioMixer::BLOCKSIZE
const uint32_t maximum = frameCount();
uint32_t clippedSize = size;
if (maximum < minimum) {
clippedSize = maximum;
} else if (clippedSize < minimum) {
clippedSize = minimum;
} else if (clippedSize > maximum) {
clippedSize = maximum;
}
// for server to read
android_atomic_release_store(clippedSize, (int32_t *)&mCblk->mBufferSizeInFrames);
// for client to read
mBufferSizeInFrames = clippedSize;
return clippedSize;
}
__attribute__((no_sanitize("integer")))
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;
}
if (flags & CBLK_DISABLED) {
ALOGV("Track disabled");
status = NOT_ENOUGH_DATA;
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;
}
// write to rear, read from front
ssize_t filled = rear - front;
// pipe should not be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
if (mIsOut) {
ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); "
"shutting down", filled, mFrameCount);
mIsShutdown = true;
status = NO_INIT;
goto end;
}
// for input, sync up on overrun
filled = 0;
cblk->u.mStreaming.mFront = rear;
(void) android_atomic_or(CBLK_OVERRUN, &cblk->mFlags);
}
// Don't allow filling pipe beyond the user settable size.
// The calculation for avail can go negative if the buffer size
// is suddenly dropped below the amount already in the buffer.
// So use a signed calculation to prevent a numeric overflow abort.
ssize_t adjustableSize = (ssize_t) getBufferSizeInFrames();
ssize_t avail = (mIsOut) ? adjustableSize - filled : filled;
if (avail < 0) {
avail = 0;
} else 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 > (size_t)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_ALWAYS_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);
status_t error = errno; // clock_gettime can affect errno
// 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 (error) {
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 = error;
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;
}
__attribute__((no_sanitize("integer")))
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) {
int32_t rear = cblk->u.mStreaming.mRear;
android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear);
} else {
int32_t front = cblk->u.mStreaming.mFront;
android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront);
}
}
void ClientProxy::binderDied()
{
audio_track_cblk_t* cblk = mCblk;
if (!(android_atomic_or(CBLK_INVALID, &cblk->mFlags) & CBLK_INVALID)) {
android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
// it seems that a FUTEX_WAKE_PRIVATE will not wake a FUTEX_WAIT, even within same process
(void) syscall(__NR_futex, &cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE,
1);
}
}
void ClientProxy::interrupt()
{
audio_track_cblk_t* cblk = mCblk;
if (!(android_atomic_or(CBLK_INTERRUPT, &cblk->mFlags) & CBLK_INTERRUPT)) {
android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
(void) syscall(__NR_futex, &cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE,
1);
}
}
__attribute__((no_sanitize("integer")))
size_t ClientProxy::getMisalignment()
{
audio_track_cblk_t* cblk = mCblk;
return (mFrameCountP2 - (mIsOut ? cblk->u.mStreaming.mRear : cblk->u.mStreaming.mFront)) &
(mFrameCountP2 - 1);
}
// ---------------------------------------------------------------------------
void AudioTrackClientProxy::flush()
{
// This works for mFrameCountP2 <= 2^30
size_t increment = mFrameCountP2 << 1;
size_t mask = increment - 1;
audio_track_cblk_t* cblk = mCblk;
// mFlush is 32 bits concatenated as [ flush_counter ] [ newfront_offset ]
// Should newFlush = cblk->u.mStreaming.mRear? Only problem is
// if you want to flush twice to the same rear location after a 32 bit wrap.
int32_t newFlush = (cblk->u.mStreaming.mRear & mask) |
((cblk->u.mStreaming.mFlush & ~mask) + increment);
android_atomic_release_store(newFlush, &cblk->u.mStreaming.mFlush);
}
bool AudioTrackClientProxy::clearStreamEndDone() {
return (android_atomic_and(~CBLK_STREAM_END_DONE, &mCblk->mFlags) & CBLK_STREAM_END_DONE) != 0;
}
bool AudioTrackClientProxy::getStreamEndDone() const {
return (mCblk->mFlags & CBLK_STREAM_END_DONE) != 0;
}
status_t AudioTrackClientProxy::waitStreamEndDone(const struct timespec *requested)
{
struct timespec total; // total elapsed time spent waiting
total.tv_sec = 0;
total.tv_nsec = 0;
audio_track_cblk_t* cblk = mCblk;
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;
}
for (;;) {
int32_t flags = android_atomic_and(~(CBLK_INTERRUPT|CBLK_STREAM_END_DONE), &cblk->mFlags);
// check for track invalidation by server, or server death detection
if (flags & CBLK_INVALID) {
ALOGV("Track invalidated");
status = DEAD_OBJECT;
goto end;
}
// a track is not supposed to underrun at this stage but consider it done
if (flags & (CBLK_STREAM_END_DONE | CBLK_DISABLED)) {
ALOGV("stream end received");
status = NO_ERROR;
goto end;
}
// check for obtainBuffer interrupted by client
if (flags & CBLK_INTERRUPT) {
ALOGV("waitStreamEndDone() interrupted by client");
status = -EINTR;
goto end;
}
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 (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_ALWAYS_FATAL("waitStreamEndDone() timeout=%d", timeout);
ts = NULL;
break;
}
int32_t old = android_atomic_and(~CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
errno = 0;
(void) syscall(__NR_futex, &cblk->mFutex,
mClientInServer ? FUTEX_WAIT_PRIVATE : FUTEX_WAIT, old & ~CBLK_FUTEX_WAKE, ts);
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
break;
default:
status = errno;
ALOGE("%s unexpected error %s", __func__, strerror(status));
goto end;
}
}
}
end:
if (requested == NULL) {
requested = &kNonBlocking;
}
return status;
}
// ---------------------------------------------------------------------------
StaticAudioTrackClientProxy::StaticAudioTrackClientProxy(audio_track_cblk_t* cblk, void *buffers,
size_t frameCount, size_t frameSize)
: AudioTrackClientProxy(cblk, buffers, frameCount, frameSize),
mMutator(&cblk->u.mStatic.mSingleStateQueue),
mPosLoopObserver(&cblk->u.mStatic.mPosLoopQueue)
{
memset(&mState, 0, sizeof(mState));
memset(&mPosLoop, 0, sizeof(mPosLoop));
}
void StaticAudioTrackClientProxy::flush()
{
LOG_ALWAYS_FATAL("static flush");
}
void StaticAudioTrackClientProxy::setLoop(size_t loopStart, size_t loopEnd, int loopCount)
{
// This can only happen on a 64-bit client
if (loopStart > UINT32_MAX || loopEnd > UINT32_MAX) {
// FIXME Should return an error status
return;
}
mState.mLoopStart = (uint32_t) loopStart;
mState.mLoopEnd = (uint32_t) loopEnd;
mState.mLoopCount = loopCount;
mState.mLoopSequence = incrementSequence(mState.mLoopSequence, mState.mPositionSequence);
// set patch-up variables until the mState is acknowledged by the ServerProxy.
// observed buffer position and loop count will freeze until then to give the
// illusion of a synchronous change.
getBufferPositionAndLoopCount(NULL, NULL);
// preserve behavior to restart at mState.mLoopStart if position exceeds mState.mLoopEnd.
if (mState.mLoopCount != 0 && mPosLoop.mBufferPosition >= mState.mLoopEnd) {
mPosLoop.mBufferPosition = mState.mLoopStart;
}
mPosLoop.mLoopCount = mState.mLoopCount;
(void) mMutator.push(mState);
}
void StaticAudioTrackClientProxy::setBufferPosition(size_t position)
{
// This can only happen on a 64-bit client
if (position > UINT32_MAX) {
// FIXME Should return an error status
return;
}
mState.mPosition = (uint32_t) position;
mState.mPositionSequence = incrementSequence(mState.mPositionSequence, mState.mLoopSequence);
// set patch-up variables until the mState is acknowledged by the ServerProxy.
// observed buffer position and loop count will freeze until then to give the
// illusion of a synchronous change.
if (mState.mLoopCount > 0) { // only check if loop count is changing
getBufferPositionAndLoopCount(NULL, NULL); // get last position
}
mPosLoop.mBufferPosition = position;
if (position >= mState.mLoopEnd) {
// no ongoing loop is possible if position is greater than loopEnd.
mPosLoop.mLoopCount = 0;
}
(void) mMutator.push(mState);
}
void StaticAudioTrackClientProxy::setBufferPositionAndLoop(size_t position, size_t loopStart,
size_t loopEnd, int loopCount)
{
setLoop(loopStart, loopEnd, loopCount);
setBufferPosition(position);
}
size_t StaticAudioTrackClientProxy::getBufferPosition()
{
getBufferPositionAndLoopCount(NULL, NULL);
return mPosLoop.mBufferPosition;
}
void StaticAudioTrackClientProxy::getBufferPositionAndLoopCount(
size_t *position, int *loopCount)
{
if (mMutator.ack() == StaticAudioTrackSingleStateQueue::SSQ_DONE) {
if (mPosLoopObserver.poll(mPosLoop)) {
; // a valid mPosLoop should be available if ackDone is true.
}
}
if (position != NULL) {
*position = mPosLoop.mBufferPosition;
}
if (loopCount != NULL) {
*loopCount = mPosLoop.mLoopCount;
}
}
// ---------------------------------------------------------------------------
ServerProxy::ServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
size_t frameSize, bool isOut, bool clientInServer)
: Proxy(cblk, buffers, frameCount, frameSize, isOut, clientInServer),
mAvailToClient(0), mFlush(0), mReleased(0), mFlushed(0)
, mTimestampMutator(&cblk->mExtendedTimestampQueue)
{
cblk->mBufferSizeInFrames = frameCount;
}
__attribute__((no_sanitize("integer")))
void ServerProxy::flushBufferIfNeeded()
{
audio_track_cblk_t* cblk = mCblk;
// The acquire_load is not really required. But since the write is a release_store in the
// client, using acquire_load here makes it easier for people to maintain the code,
// and the logic for communicating ipc variables seems somewhat standard,
// and there really isn't much penalty for 4 or 8 byte atomics.
int32_t flush = android_atomic_acquire_load(&cblk->u.mStreaming.mFlush);
if (flush != mFlush) {
ALOGV("ServerProxy::flushBufferIfNeeded() mStreaming.mFlush = 0x%x, mFlush = 0x%0x",
flush, mFlush);
int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
int32_t front = cblk->u.mStreaming.mFront;
// effectively obtain then release whatever is in the buffer
const size_t overflowBit = mFrameCountP2 << 1;
const size_t mask = overflowBit - 1;
int32_t newFront = (front & ~mask) | (flush & mask);
ssize_t filled = rear - newFront;
if (filled >= (ssize_t)overflowBit) {
// front and rear offsets span the overflow bit of the p2 mask
// so rebasing newFront on the front offset is off by the overflow bit.
// adjust newFront to match rear offset.
ALOGV("flush wrap: filled %zx >= overflowBit %zx", filled, overflowBit);
newFront += overflowBit;
filled -= overflowBit;
}
// Rather than shutting down on a corrupt flush, just treat it as a full flush
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
ALOGE("mFlush %#x -> %#x, front %#x, rear %#x, mask %#x, newFront %#x, "
"filled %zd=%#x",
mFlush, flush, front, rear,
(unsigned)mask, newFront, filled, (unsigned)filled);
newFront = rear;
}
mFlush = flush;
android_atomic_release_store(newFront, &cblk->u.mStreaming.mFront);
// There is no danger from a false positive, so err on the side of caution
if (true /*front != newFront*/) {
int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
(void) syscall(__NR_futex, &cblk->mFutex,
mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1);
}
}
mFlushed += (newFront - front) & mask;
}
}
__attribute__((no_sanitize("integer")))
status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
{
LOG_ALWAYS_FATAL_IF(buffer == NULL || buffer->mFrameCount == 0);
if (mIsShutdown) {
goto no_init;
}
{
audio_track_cblk_t* cblk = mCblk;
// compute number of frames available to write (AudioTrack) or read (AudioRecord),
// or use previous cached value from framesReady(), with added barrier if it omits.
int32_t front;
int32_t rear;
// See notes on barriers at ClientProxy::obtainBuffer()
if (mIsOut) {
flushBufferIfNeeded(); // might modify mFront
rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
front = cblk->u.mStreaming.mFront;
} else {
front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
rear = cblk->u.mStreaming.mRear;
}
ssize_t filled = rear - front;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); shutting down",
filled, mFrameCount);
mIsShutdown = true;
}
if (mIsShutdown) {
goto no_init;
}
// don't allow filling pipe beyond the nominal size
size_t availToServer;
if (mIsOut) {
availToServer = filled;
mAvailToClient = mFrameCount - filled;
} else {
availToServer = mFrameCount - filled;
mAvailToClient = filled;
}
// 'availToServer' may be non-contiguous, so return only the first contiguous chunk
size_t part1;
if (mIsOut) {
front &= mFrameCountP2 - 1;
part1 = mFrameCountP2 - front;
} else {
rear &= mFrameCountP2 - 1;
part1 = mFrameCountP2 - rear;
}
if (part1 > availToServer) {
part1 = availToServer;
}
size_t ask = buffer->mFrameCount;
if (part1 > ask) {
part1 = ask;
}
// is assignment redundant in some cases?
buffer->mFrameCount = part1;
buffer->mRaw = part1 > 0 ?
&((char *) mBuffers)[(mIsOut ? front : rear) * mFrameSize] : NULL;
buffer->mNonContig = availToServer - part1;
// After flush(), allow releaseBuffer() on a previously obtained buffer;
// see "Acknowledge any pending flush()" in audioflinger/Tracks.cpp.
if (!ackFlush) {
mUnreleased = part1;
}
return part1 > 0 ? NO_ERROR : WOULD_BLOCK;
}
no_init:
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
buffer->mNonContig = 0;
mUnreleased = 0;
return NO_INIT;
}
__attribute__((no_sanitize("integer")))
void ServerProxy::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;
if (mIsOut) {
int32_t front = cblk->u.mStreaming.mFront;
android_atomic_release_store(stepCount + front, &cblk->u.mStreaming.mFront);
} else {
int32_t rear = cblk->u.mStreaming.mRear;
android_atomic_release_store(stepCount + rear, &cblk->u.mStreaming.mRear);
}
cblk->mServer += stepCount;
mReleased += stepCount;
size_t half = mFrameCount / 2;
if (half == 0) {
half = 1;
}
size_t minimum = (size_t) cblk->mMinimum;
if (minimum == 0) {
minimum = mIsOut ? half : 1;
} else if (minimum > half) {
minimum = half;
}
// FIXME AudioRecord wakeup needs to be optimized; it currently wakes up client every time
if (!mIsOut || (mAvailToClient + stepCount >= minimum)) {
ALOGV("mAvailToClient=%zu stepCount=%zu minimum=%zu", mAvailToClient, stepCount, minimum);
int32_t old = android_atomic_or(CBLK_FUTEX_WAKE, &cblk->mFutex);
if (!(old & CBLK_FUTEX_WAKE)) {
(void) syscall(__NR_futex, &cblk->mFutex,
mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1);
}
}
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
buffer->mNonContig = 0;
}
// ---------------------------------------------------------------------------
__attribute__((no_sanitize("integer")))
size_t AudioTrackServerProxy::framesReady()
{
LOG_ALWAYS_FATAL_IF(!mIsOut);
if (mIsShutdown) {
return 0;
}
audio_track_cblk_t* cblk = mCblk;
int32_t flush = cblk->u.mStreaming.mFlush;
if (flush != mFlush) {
// FIXME should return an accurate value, but over-estimate is better than under-estimate
return mFrameCount;
}
// the acquire might not be necessary since not doing a subsequent read
int32_t rear = android_atomic_acquire_load(&cblk->u.mStreaming.mRear);
ssize_t filled = rear - cblk->u.mStreaming.mFront;
// pipe should not already be overfull
if (!(0 <= filled && (size_t) filled <= mFrameCount)) {
ALOGE("Shared memory control block is corrupt (filled=%zd, mFrameCount=%zu); shutting down",
filled, mFrameCount);
mIsShutdown = true;
return 0;
}
// cache this value for later use by obtainBuffer(), with added barrier
// and racy if called by normal mixer thread
// ignores flush(), so framesReady() may report a larger mFrameCount than obtainBuffer()
return filled;
}
bool AudioTrackServerProxy::setStreamEndDone() {
audio_track_cblk_t* cblk = mCblk;
bool old =
(android_atomic_or(CBLK_STREAM_END_DONE, &cblk->mFlags) & CBLK_STREAM_END_DONE) != 0;
if (!old) {
(void) syscall(__NR_futex, &cblk->mFutex, mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE,
1);
}
return old;
}
void AudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount)
{
audio_track_cblk_t* cblk = mCblk;
if (frameCount > 0) {
cblk->u.mStreaming.mUnderrunFrames += frameCount;
if (!mUnderrunning) { // start of underrun?
mUnderrunCount++;
cblk->u.mStreaming.mUnderrunCount = mUnderrunCount;
mUnderrunning = true;
ALOGV("tallyUnderrunFrames(%3u) at uf = %u, bump mUnderrunCount = %u",
frameCount, cblk->u.mStreaming.mUnderrunFrames, mUnderrunCount);
}
// FIXME also wake futex so that underrun is noticed more quickly
(void) android_atomic_or(CBLK_UNDERRUN, &cblk->mFlags);
} else {
ALOGV_IF(mUnderrunning,
"tallyUnderrunFrames(%3u) at uf = %u, underrun finished",
frameCount, cblk->u.mStreaming.mUnderrunFrames);
mUnderrunning = false; // so we can detect the next edge
}
}
AudioPlaybackRate AudioTrackServerProxy::getPlaybackRate()
{ // do not call from multiple threads without holding lock
mPlaybackRateObserver.poll(mPlaybackRate);
return mPlaybackRate;
}
// ---------------------------------------------------------------------------
StaticAudioTrackServerProxy::StaticAudioTrackServerProxy(audio_track_cblk_t* cblk, void *buffers,
size_t frameCount, size_t frameSize)
: AudioTrackServerProxy(cblk, buffers, frameCount, frameSize),
mObserver(&cblk->u.mStatic.mSingleStateQueue),
mPosLoopMutator(&cblk->u.mStatic.mPosLoopQueue),
mFramesReadySafe(frameCount), mFramesReady(frameCount),
mFramesReadyIsCalledByMultipleThreads(false)
{
memset(&mState, 0, sizeof(mState));
}
void StaticAudioTrackServerProxy::framesReadyIsCalledByMultipleThreads()
{
mFramesReadyIsCalledByMultipleThreads = true;
}
size_t StaticAudioTrackServerProxy::framesReady()
{
// Can't call pollPosition() from multiple threads.
if (!mFramesReadyIsCalledByMultipleThreads) {
(void) pollPosition();
}
return mFramesReadySafe;
}
status_t StaticAudioTrackServerProxy::updateStateWithLoop(
StaticAudioTrackState *localState, const StaticAudioTrackState &update) const
{
if (localState->mLoopSequence != update.mLoopSequence) {
bool valid = false;
const size_t loopStart = update.mLoopStart;
const size_t loopEnd = update.mLoopEnd;
size_t position = localState->mPosition;
if (update.mLoopCount == 0) {
valid = true;
} else if (update.mLoopCount >= -1) {
if (loopStart < loopEnd && loopEnd <= mFrameCount &&
loopEnd - loopStart >= MIN_LOOP) {
// If the current position is greater than the end of the loop
// we "wrap" to the loop start. This might cause an audible pop.
if (position >= loopEnd) {
position = loopStart;
}
valid = true;
}
}
if (!valid || position > mFrameCount) {
return NO_INIT;
}
localState->mPosition = position;
localState->mLoopCount = update.mLoopCount;
localState->mLoopEnd = loopEnd;
localState->mLoopStart = loopStart;
localState->mLoopSequence = update.mLoopSequence;
}
return OK;
}
status_t StaticAudioTrackServerProxy::updateStateWithPosition(
StaticAudioTrackState *localState, const StaticAudioTrackState &update) const
{
if (localState->mPositionSequence != update.mPositionSequence) {
if (update.mPosition > mFrameCount) {
return NO_INIT;
} else if (localState->mLoopCount != 0 && update.mPosition >= localState->mLoopEnd) {
localState->mLoopCount = 0; // disable loop count if position is beyond loop end.
}
localState->mPosition = update.mPosition;
localState->mPositionSequence = update.mPositionSequence;
}
return OK;
}
ssize_t StaticAudioTrackServerProxy::pollPosition()
{
StaticAudioTrackState state;
if (mObserver.poll(state)) {
StaticAudioTrackState trystate = mState;
bool result;
const int32_t diffSeq = (int32_t) state.mLoopSequence - (int32_t) state.mPositionSequence;
if (diffSeq < 0) {
result = updateStateWithLoop(&trystate, state) == OK &&
updateStateWithPosition(&trystate, state) == OK;
} else {
result = updateStateWithPosition(&trystate, state) == OK &&
updateStateWithLoop(&trystate, state) == OK;
}
if (!result) {
mObserver.done();
// caution: no update occurs so server state will be inconsistent with client state.
ALOGE("%s client pushed an invalid state, shutting down", __func__);
mIsShutdown = true;
return (ssize_t) NO_INIT;
}
mState = trystate;
if (mState.mLoopCount == -1) {
mFramesReady = INT64_MAX;
} else if (mState.mLoopCount == 0) {
mFramesReady = mFrameCount - mState.mPosition;
} else if (mState.mLoopCount > 0) {
// TODO: Later consider fixing overflow, but does not seem needed now
// as will not overflow if loopStart and loopEnd are Java "ints".
mFramesReady = int64_t(mState.mLoopCount) * (mState.mLoopEnd - mState.mLoopStart)
+ mFrameCount - mState.mPosition;
}
mFramesReadySafe = clampToSize(mFramesReady);
// This may overflow, but client is not supposed to rely on it
StaticAudioTrackPosLoop posLoop;
posLoop.mLoopCount = (int32_t) mState.mLoopCount;
posLoop.mBufferPosition = (uint32_t) mState.mPosition;
mPosLoopMutator.push(posLoop);
mObserver.done(); // safe to read mStatic variables.
}
return (ssize_t) mState.mPosition;
}
status_t StaticAudioTrackServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
{
if (mIsShutdown) {
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
buffer->mNonContig = 0;
mUnreleased = 0;
return NO_INIT;
}
ssize_t positionOrStatus = pollPosition();
if (positionOrStatus < 0) {
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
buffer->mNonContig = 0;
mUnreleased = 0;
return (status_t) positionOrStatus;
}
size_t position = (size_t) positionOrStatus;
size_t end = mState.mLoopCount != 0 ? mState.mLoopEnd : mFrameCount;
size_t avail;
if (position < end) {
avail = end - position;
size_t wanted = buffer->mFrameCount;
if (avail < wanted) {
buffer->mFrameCount = avail;
} else {
avail = wanted;
}
buffer->mRaw = &((char *) mBuffers)[position * mFrameSize];
} else {
avail = 0;
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
}
// As mFramesReady is the total remaining frames in the static audio track,
// it is always larger or equal to avail.
LOG_ALWAYS_FATAL_IF(mFramesReady < (int64_t) avail);
buffer->mNonContig = mFramesReady == INT64_MAX ? SIZE_MAX : clampToSize(mFramesReady - avail);
if (!ackFlush) {
mUnreleased = avail;
}
return NO_ERROR;
}
void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
{
size_t stepCount = buffer->mFrameCount;
LOG_ALWAYS_FATAL_IF(!((int64_t) stepCount <= mFramesReady));
LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased));
if (stepCount == 0) {
// prevent accidental re-use of buffer
buffer->mRaw = NULL;
buffer->mNonContig = 0;
return;
}
mUnreleased -= stepCount;
audio_track_cblk_t* cblk = mCblk;
size_t position = mState.mPosition;
size_t newPosition = position + stepCount;
int32_t setFlags = 0;
if (!(position <= newPosition && newPosition <= mFrameCount)) {
ALOGW("%s newPosition %zu outside [%zu, %zu]", __func__, newPosition, position,
mFrameCount);
newPosition = mFrameCount;
} else if (mState.mLoopCount != 0 && newPosition == mState.mLoopEnd) {
newPosition = mState.mLoopStart;
if (mState.mLoopCount == -1 || --mState.mLoopCount != 0) {
setFlags = CBLK_LOOP_CYCLE;
} else {
setFlags = CBLK_LOOP_FINAL;
}
}
if (newPosition == mFrameCount) {
setFlags |= CBLK_BUFFER_END;
}
mState.mPosition = newPosition;
if (mFramesReady != INT64_MAX) {
mFramesReady -= stepCount;
}
mFramesReadySafe = clampToSize(mFramesReady);
cblk->mServer += stepCount;
mReleased += stepCount;
// This may overflow, but client is not supposed to rely on it
StaticAudioTrackPosLoop posLoop;
posLoop.mBufferPosition = mState.mPosition;
posLoop.mLoopCount = mState.mLoopCount;
mPosLoopMutator.push(posLoop);
if (setFlags != 0) {
(void) android_atomic_or(setFlags, &cblk->mFlags);
// this would be a good place to wake a futex
}
buffer->mFrameCount = 0;
buffer->mRaw = NULL;
buffer->mNonContig = 0;
}
void StaticAudioTrackServerProxy::tallyUnderrunFrames(uint32_t frameCount)
{
// Unlike AudioTrackServerProxy::tallyUnderrunFrames() used for streaming tracks,
// we don't have a location to count underrun frames. The underrun frame counter
// only exists in AudioTrackSharedStreaming. Fortunately, underruns are not
// possible for static buffer tracks other than at end of buffer, so this is not a loss.
// FIXME also wake futex so that underrun is noticed more quickly
if (frameCount > 0) {
(void) android_atomic_or(CBLK_UNDERRUN, &mCblk->mFlags);
}
}
// ---------------------------------------------------------------------------
} // namespace android