Merge "AudioFlinger::Thread ensure proper add / sub" into qt-dev (65443895) · Commits · e / os / android_frameworks_av

media/libaudioclient/AudioTrackShared.cpp

+8 −25

Original line number	Diff line number	Diff line
		@@ -20,30 +20,13 @@
		#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 {

		// TODO: consider pulling this into a shared header.
		// safe_sub_overflow is used ensure that subtraction occurs in the same native type
		// with proper 2's complement overflow. Without calling this function, it is possible,
		// for example, that optimizing compilers may elect to treat 32 bit subtraction
		// as 64 bit subtraction when storing into a 64 bit destination as integer overflow is
		// technically undefined.
		template<typename T,
		typename U,
		typename = std::enable_if_t<std::is_same<std::decay_t<T>,
		std::decay_t<U>>{}>>
		// ensure arguments are same type (ignoring volatile, which is used in cblk variables).
		auto safe_sub_overflow(const T& a, const U& b) {
		std::decay_t<T> result;
		(void)__builtin_sub_overflow(a, b, &result);
		// note if __builtin_sub_overflow returns true, an overflow occurred.
		return result;
		}

		// used to clamp a value to size_t. TODO: move to another file.
		template <typename T>
		size_t clampToSize(T x) {
		@@ -204,7 +187,7 @@ status_t ClientProxy::obtainBuffer(Buffer* buffer, const struct timespec *reques
		front = cblk->u.mStreaming.mFront;
		}
		// write to rear, read from front
		ssize_t filled = safe_sub_overflow(rear, 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) {
		@@ -702,7 +685,7 @@ void ServerProxy::flushBufferIfNeeded()
		const size_t overflowBit = mFrameCountP2 << 1;
		const size_t mask = overflowBit - 1;
		int32_t newFront = (front & ~mask) \| (flush & mask);
		ssize_t filled = safe_sub_overflow(rear, newFront);
		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.
		@@ -744,7 +727,7 @@ int32_t AudioTrackServerProxy::getRear() const
		const size_t overflowBit = mFrameCountP2 << 1;
		const size_t mask = overflowBit - 1;
		int32_t newRear = (rear & ~mask) \| (stop & mask);
		ssize_t filled = safe_sub_overflow(newRear, front);
		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
		@@ -796,7 +779,7 @@ status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
		front = android_atomic_acquire_load(&cblk->u.mStreaming.mFront);
		rear = cblk->u.mStreaming.mRear;
		}
		ssize_t filled = safe_sub_overflow(rear, front);
		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",
		@@ -923,7 +906,7 @@ size_t AudioTrackServerProxy::framesReady()
		return mFrameCount;
		}
		const int32_t rear = getRear();
		ssize_t filled = safe_sub_overflow(rear, cblk->u.mStreaming.mFront);
		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",
		@@ -949,7 +932,7 @@ size_t AudioTrackServerProxy::framesReadySafe() const
		return mFrameCount;
		}
		const int32_t rear = getRear();
		const ssize_t filled = safe_sub_overflow(rear, cblk->u.mStreaming.mFront);
		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.
		}
		@@ -1259,7 +1242,7 @@ size_t AudioRecordServerProxy::framesReadySafe() const
		}
		const int32_t front = android_atomic_acquire_load(&mCblk->u.mStreaming.mFront);
		const int32_t rear = mCblk->u.mStreaming.mRear;
		const ssize_t filled = safe_sub_overflow(rear, front);
		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.
		}

services/audioflinger/Threads.cpp

+7 −5

Original line number	Diff line number	Diff line
		@@ -44,6 +44,7 @@
		#include <audio_utils/primitives.h>
		#include <audio_utils/format.h>
		#include <audio_utils/minifloat.h>
		#include <audio_utils/safe_math.h>
		#include <system/audio_effects/effect_ns.h>
		#include <system/audio_effects/effect_aec.h>
		#include <system/audio.h>
		@@ -7904,7 +7905,7 @@ void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
		RecordThread recordThread = (RecordThread ) threadBase.get();
		const int32_t rear = recordThread->mRsmpInRear;
		const int32_t front = mRsmpInFront;
		const ssize_t filled = rear - front;
		const ssize_t filled = audio_utils::safe_sub_overflow(rear, front);

		size_t framesIn;
		bool overrun = false;
		@@ -7918,7 +7919,8 @@ void AudioFlinger::RecordThread::ResamplerBufferProvider::sync(
		} else {
		// client is not keeping up with server, but give it latest data
		framesIn = recordThread->mRsmpInFrames;
		mRsmpInFront = /* front = */ rear - framesIn;
		mRsmpInFront = /* front = */ audio_utils::safe_sub_overflow(
		rear, static_cast<int32_t>(framesIn));
		overrun = true;
		}
		if (framesAvailable != NULL) {
		@@ -7942,7 +7944,7 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
		RecordThread recordThread = (RecordThread ) threadBase.get();
		int32_t rear = recordThread->mRsmpInRear;
		int32_t front = mRsmpInFront;
		ssize_t filled = rear - front;
		ssize_t filled = audio_utils::safe_sub_overflow(rear, front);
		// FIXME should not be P2 (don't want to increase latency)
		// FIXME if client not keeping up, discard
		LOG_ALWAYS_FATAL_IF(!(0 <= filled && (size_t) filled <= recordThread->mRsmpInFrames));
		@@ -7975,13 +7977,13 @@ status_t AudioFlinger::RecordThread::ResamplerBufferProvider::getNextBuffer(
		void AudioFlinger::RecordThread::ResamplerBufferProvider::releaseBuffer(
		AudioBufferProvider::Buffer* buffer)
		{
		size_t stepCount = buffer->frameCount;
		int32_t stepCount = static_cast<int32_t>(buffer->frameCount);
		if (stepCount == 0) {
		return;
		}
		ALOG_ASSERT(stepCount <= mRsmpInUnrel);
		mRsmpInUnrel -= stepCount;
		mRsmpInFront += stepCount;
		mRsmpInFront = audio_utils::safe_add_overflow(mRsmpInFront, stepCount);
		buffer->raw = NULL;
		buffer->frameCount = 0;
		}