Merge "audio flinger: fix wraparound issues in AudioRecord shared audi history... (86fec3ec) · Commits · e / os / android_frameworks_av

services/audioflinger/RecordTracks.h

+3 −3

Original line number	Diff line number	Diff line
		@@ -70,7 +70,7 @@ public:
		audio_input_flags_t flags,
		track_type type,
		audio_port_handle_t portId = AUDIO_PORT_HANDLE_NONE,
		int64_t startTimeMs = -1);
		int32_t startFrames = -1);
		virtual ~RecordTrack();
		virtual status_t initCheck() const;

		@@ -110,7 +110,7 @@ public:
		status_t setPreferredMicrophoneFieldDimension(float zoom);
		status_t shareAudioHistory(const std::string& sharedAudioPackageName,
		int64_t sharedAudioStartMs);
		int64_t startTimeMs() { return mStartTimeMs; }
		int32_t startFrames() { return mStartFrames; }

		static bool checkServerLatencySupported(
		audio_format_t format, audio_input_flags_t flags) {
		@@ -152,7 +152,7 @@ private:
		// used to enforce OP_RECORD_AUDIO
		sp<OpRecordAudioMonitor> mOpRecordAudioMonitor;
		std::string mSharedAudioPackageName = {};
		int64_t mStartTimeMs = -1;
		int32_t mStartFrames = -1;
		};

		// playback track, used by PatchPanel

services/audioflinger/Threads.cpp

+43 −23

Original line number	Diff line number	Diff line
		@@ -7934,18 +7934,18 @@ sp<AudioFlinger::RecordThread::RecordTrack> AudioFlinger::RecordThread::createRe

		{ // scope for mLock
		Mutex::Autolock _l(mLock);
		long startTimeMs = -1;
		int32_t startFrames = -1;
		if (!mSharedAudioPackageName.empty()
		&& mSharedAudioPackageName == checkedIdentity.packageName
		&& mSharedAudioSessionId == sessionId
		&& captureHotwordAllowed(checkedIdentity)) {
		startTimeMs = mSharedAudioStartMs;
		startFrames = mSharedAudioStartFrames;
		}

		track = new RecordTrack(this, client, attr, sampleRate,
		format, channelMask, frameCount,
		nullptr /* buffer /, (size_t)0 / bufferSize */, sessionId, creatorPid,
		checkedIdentity, *flags, TrackBase::TYPE_DEFAULT, portId, startTimeMs);
		checkedIdentity, *flags, TrackBase::TYPE_DEFAULT, portId, startFrames);

		lStatus = track->initCheck();
		if (lStatus != NO_ERROR) {
		@@ -8202,17 +8202,32 @@ status_t AudioFlinger::RecordThread::shareAudioHistory_l(
		if ((hasAudioSession_l(sharedSessionId) & ThreadBase::TRACK_SESSION) == 0) {
		return BAD_VALUE;
		}
		if (sharedAudioStartMs < 0 \|\| sharedAudioStartMs * mSampleRate / 1000 > mRsmpInRear) {

		if (sharedAudioStartMs < 0
		\|\| sharedAudioStartMs > INT64_MAX / mSampleRate) {
		return BAD_VALUE;
		}

		// Current implementation of the input resampling buffer wraps around indexes at 32 bit.
		// As we cannot detect more than one wraparound, only accept values up current write position
		// after one wraparound
		// We assume recent wraparounds on mRsmpInRear only given it is unlikely that the requesting
		// app waits several hours after the start time was computed.
		const int64_t sharedAudioStartFrames = sharedAudioStartMs * mSampleRate / 1000;
		const int32_t sharedOffset = audio_utils::safe_sub_overflow(mRsmpInRear,
		(int32_t)sharedAudioStartFrames);
		if (sharedOffset < 0
		\|\| sharedOffset > mRsmpInFrames) {
		return BAD_VALUE;
		}

		mSharedAudioPackageName = sharedAudioPackageName;
		if (mSharedAudioPackageName.empty()) {
		mSharedAudioSessionId = AUDIO_SESSION_NONE;
		mSharedAudioStartMs = -1;
		mSharedAudioStartFrames = -1;
		} else {
		mSharedAudioSessionId = sharedSessionId;
		mSharedAudioStartMs = sharedAudioStartMs;
		mSharedAudioStartFrames = (int32_t)sharedAudioStartFrames;
		}
		return NO_ERROR;
		}
		@@ -8359,14 +8374,14 @@ void AudioFlinger::RecordThread::ResamplerBufferProvider::reset()
		mRsmpInUnrel = 0;
		const int32_t rear = recordThread->mRsmpInRear;
		ssize_t deltaFrames = 0;
		if (mRecordTrack->startTimeMs() >= 0) {
		int32_t startFrames = mRecordTrack->startTimeMs() * recordThread->sampleRate() / 1000;
		// start frame has to be in the past
		//TODO: b/185972521 fix in case rear or startFrames wrap around
		if (startFrames > rear) {
		startFrames = rear;
		}
		if (mRecordTrack->startFrames() >= 0) {
		int32_t startFrames = mRecordTrack->startFrames();
		// Accept a recent wraparound of mRsmpInRear
		if (startFrames <= rear) {
		deltaFrames = rear - startFrames;
		} else {
		deltaFrames = (int32_t)((int64_t)rear + UINT32_MAX + 1 - startFrames);
		}
		// start frame cannot be further in the past than start of resampling buffer
		if ((size_t) deltaFrames > recordThread->mRsmpInFrames) {
		deltaFrames = recordThread->mRsmpInFrames;
		@@ -8828,17 +8843,22 @@ int32_t AudioFlinger::RecordThread::getOldestFront_l()
		if (mTracks.size() == 0) {
		return 0;
		}
		//TODO: b/185972521 fix in case of wrap around on one track:
		// want the max(rear - front) for all tracks.
		int32_t front = INT_MAX;
		int32_t oldestFront = mRsmpInRear;
		int32_t maxFilled = 0;
		for (size_t i = 0; i < mTracks.size(); i++) {
		front = std::min(front, mTracks[i]->mResamplerBufferProvider->getFront());
		int32_t front = mTracks[i]->mResamplerBufferProvider->getFront();
		int32_t filled;
		if (front <= mRsmpInRear) {
		filled = mRsmpInRear - front;
		} else {
		filled = (int32_t)((int64_t)mRsmpInRear + UINT32_MAX + 1 - front);
		}
		if (filled > maxFilled) {
		oldestFront = front;
		maxFilled = filled;
		}
		// discard any audio past the buffer size
		if (audio_utils::safe_add_overflow(front, (int32_t)mRsmpInFrames) < mRsmpInRear) {
		front = audio_utils::safe_sub_overflow(mRsmpInRear, (int32_t)mRsmpInFrames);
		}
		return front;
		return oldestFront;
		}

		void AudioFlinger::RecordThread::updateFronts_l(int32_t offset)

services/audioflinger/Threads.h

+1 −1

Original line number	Diff line number	Diff line
		@@ -1863,7 +1863,7 @@ private:
		DeviceDescriptorBaseVector mOutDevices;

		std::string mSharedAudioPackageName = {};
		long mSharedAudioStartMs = 0;
		int32_t mSharedAudioStartFrames = -1;
		audio_session_t mSharedAudioSessionId = AUDIO_SESSION_NONE;
		};

services/audioflinger/Tracks.cpp

+2 −2

Original line number	Diff line number	Diff line
		@@ -2378,7 +2378,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
		audio_input_flags_t flags,
		track_type type,
		audio_port_handle_t portId,
		int64_t startTimeMs)
		int32_t startFrames)
		: TrackBase(thread, client, attr, sampleRate, format,
		channelMask, frameCount, buffer, bufferSize, sessionId,
		creatorPid,
		@@ -2396,7 +2396,7 @@ AudioFlinger::RecordThread::RecordTrack::RecordTrack(
		mFlags(flags),
		mSilenced(false),
		mOpRecordAudioMonitor(OpRecordAudioMonitor::createIfNeeded(identity, attr)),
		mStartTimeMs(startTimeMs)
		mStartFrames(startFrames)
		{
		if (mCblk == NULL) {
		return;