| /* |
| * 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 <atomic> |
| #include <android-base/macros.h> |
| #include <private/media/AudioTrackShared.h> |
| #include <utils/Log.h> |
| #include <audio_utils/safe_math.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; |
| } |
| |
| // compile-time safe atomics. TODO: update all methods to use it |
| template <typename T> |
| T android_atomic_load(const volatile T* addr) { |
| static_assert(sizeof(T) == sizeof(std::atomic<T>)); // no extra sync data required. |
| static_assert(std::atomic<T>::is_always_lock_free); // no hash lock somewhere. |
| return atomic_load((std::atomic<T>*)addr); // memory_order_seq_cst |
| } |
| |
| template <typename T> |
| void android_atomic_store(const volatile T* addr, T value) { |
| static_assert(sizeof(T) == sizeof(std::atomic<T>)); // no extra sync data required. |
| static_assert(std::atomic<T>::is_always_lock_free); // no hash lock somewhere. |
| atomic_store((std::atomic<T>*)addr, value); // memory_order_seq_cst |
| } |
| |
| // 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. |
| __attribute__((no_sanitize("integer"))) |
| 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) |
| , mStartThresholdInFrames(0) // filled in by the server. |
| , 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) |
| { |
| } |
| |
| uint32_t Proxy::getStartThresholdInFrames() const |
| { |
| const uint32_t startThresholdInFrames = |
| android_atomic_load(&mCblk->mStartThresholdInFrames); |
| if (startThresholdInFrames == 0 || startThresholdInFrames > mFrameCount) { |
| ALOGD("%s: startThresholdInFrames %u not between 1 and frameCount %zu, " |
| "setting to frameCount", |
| __func__, startThresholdInFrames, mFrameCount); |
| return mFrameCount; |
| } |
| return startThresholdInFrames; |
| } |
| |
| uint32_t Proxy::setStartThresholdInFrames(uint32_t startThresholdInFrames) |
| { |
| const uint32_t actual = std::min((size_t)startThresholdInFrames, frameCount()); |
| android_atomic_store(&mCblk->mStartThresholdInFrames, actual); |
| return actual; |
| } |
| |
| // --------------------------------------------------------------------------- |
| |
| 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, |
| "%s: null or zero frame buffer, buffer:%p", __func__, buffer); |
| 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 = audio_utils::safe_sub_overflow(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; |
| } |
| FALLTHROUGH_INTENDED; |
| 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; |
| // Use auto instead of long to avoid the google-runtime-int warning. |
| auto 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), |
| "%s: mUnreleased out of range, " |
| "!(stepCount:%zu <= mUnreleased:%zu <= mFrameCount:%zu), BufferSizeInFrames:%u", |
| __func__, stepCount, mUnreleased, mFrameCount, getBufferSizeInFrames()); |
| 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, |
| INT_MAX); |
| } |
| } |
| |
| 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, |
| INT_MAX); |
| } |
| } |
| |
| __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() |
| { |
| sendStreamingFlushStop(true /* flush */); |
| } |
| |
| void AudioTrackClientProxy::stop() |
| { |
| sendStreamingFlushStop(false /* flush */); |
| } |
| |
| // Sets the client-written mFlush and mStop positions, which control server behavior. |
| // |
| // @param flush indicates whether the operation is a flush or stop. |
| // A client stop sets mStop to the current write position; |
| // the server will not read past this point until start() or subsequent flush(). |
| // A client flush sets both mStop and mFlush to the current write position. |
| // This advances the server read limit (if previously set) and on the next |
| // server read advances the server read position to this limit. |
| // |
| void AudioTrackClientProxy::sendStreamingFlushStop(bool flush) |
| { |
| // TODO: Replace this by 64 bit counters - avoids wrap complication. |
| // This works for mFrameCountP2 <= 2^30 |
| // 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. |
| |
| const size_t increment = mFrameCountP2 << 1; |
| const size_t mask = increment - 1; |
| // No need for client atomic synchronization on mRear, mStop, mFlush |
| // as AudioTrack client only read/writes to them under client lock. Server only reads. |
| const int32_t rearMasked = mCblk->u.mStreaming.mRear & mask; |
| |
| // update stop before flush so that the server front |
| // never advances beyond a (potential) previous stop's rear limit. |
| int32_t stopBits; // the following add can overflow |
| __builtin_add_overflow(mCblk->u.mStreaming.mStop & ~mask, increment, &stopBits); |
| android_atomic_release_store(rearMasked | stopBits, &mCblk->u.mStreaming.mStop); |
| |
| if (flush) { |
| int32_t flushBits; // the following add can overflow |
| __builtin_add_overflow(mCblk->u.mStreaming.mFlush & ~mask, increment, &flushBits); |
| android_atomic_release_store(rearMasked | flushBits, &mCblk->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 |
| struct timespec before; |
| bool beforeIsValid = false; |
| 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; |
| } |
| FALLTHROUGH_INTENDED; |
| 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)) { |
| if (!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 |
| { |
| struct timespec after; |
| clock_gettime(CLOCK_MONOTONIC, &after); |
| total.tv_sec += after.tv_sec - before.tv_sec; |
| // Use auto instead of long to avoid the google-runtime-int warning. |
| auto 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; |
| } |
| 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 |
| break; |
| default: |
| status = error; |
| 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::stop() |
| { |
| ; // no special handling required for static tracks. |
| } |
| |
| 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; |
| cblk->mStartThresholdInFrames = 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); |
| // shouldn't matter, but for range safety use mRear instead of getRear(). |
| 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 = audio_utils::safe_sub_overflow(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, INT_MAX); |
| } |
| } |
| mFlushed += (newFront - front) & mask; |
| } |
| } |
| |
| __attribute__((no_sanitize("integer"))) |
| int32_t AudioTrackServerProxy::getRear() const |
| { |
| const int32_t stop = android_atomic_acquire_load(&mCblk->u.mStreaming.mStop); |
| const int32_t rear = android_atomic_acquire_load(&mCblk->u.mStreaming.mRear); |
| const int32_t stopLast = mStopLast.load(std::memory_order_acquire); |
| if (stop != stopLast) { |
| const int32_t front = mCblk->u.mStreaming.mFront; |
| const size_t overflowBit = mFrameCountP2 << 1; |
| const size_t mask = overflowBit - 1; |
| int32_t newRear = (rear & ~mask) | (stop & mask); |
| ssize_t filled = audio_utils::safe_sub_overflow(newRear, front); |
| // overflowBit is unsigned, so cast to signed for comparison. |
| if (filled >= (ssize_t)overflowBit) { |
| // front and rear offsets span the overflow bit of the p2 mask |
| // so rebasing newRear on the rear offset is off by the overflow bit. |
| ALOGV("stop wrap: filled %zx >= overflowBit %zx", filled, overflowBit); |
| newRear -= overflowBit; |
| filled -= overflowBit; |
| } |
| if (0 <= filled && (size_t) filled <= mFrameCount) { |
| // we're stopped, return the stop level as newRear |
| return newRear; |
| } |
| |
| // A corrupt stop. Log error and ignore. |
| ALOGE("mStopLast %#x -> stop %#x, front %#x, rear %#x, mask %#x, newRear %#x, " |
| "filled %zd=%#x", |
| stopLast, stop, front, rear, |
| (unsigned)mask, newRear, filled, (unsigned)filled); |
| // Don't reset mStopLast as this is const. |
| } |
| return rear; |
| } |
| |
| void AudioTrackServerProxy::start() |
| { |
| mStopLast = android_atomic_acquire_load(&mCblk->u.mStreaming.mStop); |
| } |
| |
| __attribute__((no_sanitize("integer"))) |
| status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush) |
| { |
| LOG_ALWAYS_FATAL_IF(buffer == NULL || buffer->mFrameCount == 0, |
| "%s: null or zero frame buffer, buffer:%p", __func__, buffer); |
| 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 = getRear(); |
| front = cblk->u.mStreaming.mFront; |
| } else { |
| front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront); |
| rear = cblk->u.mStreaming.mRear; |
| } |
| ssize_t filled = audio_utils::safe_sub_overflow(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), |
| "%s: mUnreleased out of range, " |
| "!(stepCount:%zu <= mUnreleased:%zu <= mFrameCount:%zu)", |
| __func__, stepCount, 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, INT_MAX); |
| } |
| } |
| |
| 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; |
| |
| flushBufferIfNeeded(); |
| |
| const int32_t rear = getRear(); |
| ssize_t filled = audio_utils::safe_sub_overflow(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; |
| } |
| |
| __attribute__((no_sanitize("integer"))) |
| size_t AudioTrackServerProxy::framesReadySafe() const |
| { |
| if (mIsShutdown) { |
| return 0; |
| } |
| const audio_track_cblk_t* cblk = mCblk; |
| const int32_t flush = android_atomic_acquire_load(&cblk->u.mStreaming.mFlush); |
| if (flush != mFlush) { |
| return mFrameCount; |
| } |
| const int32_t rear = getRear(); |
| const ssize_t filled = audio_utils::safe_sub_overflow(rear, cblk->u.mStreaming.mFront); |
| if (!(0 <= filled && (size_t) filled <= mFrameCount)) { |
| return 0; // error condition, silently return 0. |
| } |
| 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; |
| } |
| |
| __attribute__((no_sanitize("integer"))) |
| 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, uint32_t sampleRate) |
| : AudioTrackServerProxy(cblk, buffers, frameCount, frameSize, false /*clientInServer*/, |
| sampleRate), |
| 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; |
| } |
| |
| size_t StaticAudioTrackServerProxy::framesReadySafe() const |
| { |
| 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; |
| } |
| |
| __attribute__((no_sanitize("integer"))) |
| 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, |
| "%s: mFramesReady out of range, mFramesReady:%lld < avail:%zu", |
| __func__, (long long)mFramesReady, avail); |
| buffer->mNonContig = mFramesReady == INT64_MAX ? SIZE_MAX : clampToSize(mFramesReady - avail); |
| if (!ackFlush) { |
| mUnreleased = avail; |
| } |
| return NO_ERROR; |
| } |
| |
| __attribute__((no_sanitize("integer"))) |
| void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer) |
| { |
| size_t stepCount = buffer->mFrameCount; |
| LOG_ALWAYS_FATAL_IF(!((int64_t) stepCount <= mFramesReady), |
| "%s: stepCount out of range, " |
| "!(stepCount:%zu <= mFramesReady:%lld)", |
| __func__, stepCount, (long long)mFramesReady); |
| LOG_ALWAYS_FATAL_IF(!(stepCount <= mUnreleased), |
| "%s: stepCount out of range, " |
| "!(stepCount:%zu <= mUnreleased:%zu)", |
| __func__, 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); |
| } |
| } |
| |
| int32_t StaticAudioTrackServerProxy::getRear() const |
| { |
| LOG_ALWAYS_FATAL("getRear() not permitted for static tracks"); |
| return 0; |
| } |
| |
| __attribute__((no_sanitize("integer"))) |
| size_t AudioRecordServerProxy::framesReadySafe() const |
| { |
| if (mIsShutdown) { |
| return 0; |
| } |
| const int32_t front = android_atomic_acquire_load(&mCblk->u.mStreaming.mFront); |
| const int32_t rear = mCblk->u.mStreaming.mRear; |
| const ssize_t filled = audio_utils::safe_sub_overflow(rear, front); |
| if (!(0 <= filled && (size_t) filled <= mFrameCount)) { |
| return 0; // error condition, silently return 0. |
| } |
| return filled; |
| } |
| |
| // --------------------------------------------------------------------------- |
| |
| } // namespace android |