Merge "Add AudioRecord timestamps" into nyc-dev (2c7994c7) · Commits · e / os / android_frameworks_av

include/media/AudioRecord.h

+17 −0

Original line number	Diff line number	Diff line
		@@ -19,6 +19,7 @@

		#include <cutils/sched_policy.h>
		#include <media/AudioSystem.h>
		#include <media/AudioTimestamp.h>
		#include <media/IAudioRecord.h>
		#include <media/Modulo.h>
		#include <utils/threads.h>
		@@ -314,6 +315,17 @@ public:
		*/
		status_t getPosition(uint32_t *position) const;

		/* Return the record timestamp.
		*
		* Parameters:
		* timestamp: A pointer to the timestamp to be filled.
		*
		* Returned status (from utils/Errors.h) can be:
		* - NO_ERROR: successful operation
		* - BAD_VALUE: timestamp is NULL
		*/
		status_t getTimestamp(ExtendedTimestamp *timestamp);

		/* Returns a handle on the audio input used by this AudioRecord.
		*
		* Parameters:
		@@ -571,6 +583,11 @@ private:
		size_t mReqFrameCount; // frame count to request the first or next time
		// a new IAudioRecord is needed, non-decreasing

		int64_t mFramesRead; // total frames read. reset to zero after
		// the start() following stop(). It is not
		// changed after restoring the track.
		int64_t mFramesReadServerOffset; // An offset to server frames read due to
		// restoring AudioRecord, or stop/start.
		// constant after constructor or set()
		uint32_t mSampleRate;
		audio_format_t mFormat;

include/media/AudioTimestamp.h

+81 −0

Original line number	Diff line number	Diff line
		@@ -17,6 +17,8 @@
		#ifndef ANDROID_AUDIO_TIMESTAMP_H
		#define ANDROID_AUDIO_TIMESTAMP_H

		#include <string>
		#include <sstream>
		#include <time.h>

		namespace android {
		@@ -32,6 +34,85 @@ public:
		struct timespec mTime; // corresponding CLOCK_MONOTONIC when frame is expected to present
		};

		struct ExtendedTimestamp {
		enum Location {
		LOCATION_CLIENT, // timestamp of last read frame from client-server track buffer
		LOCATION_SERVER, // timestamp of newest frame from client-server track buffer
		LOCATION_KERNEL, // timestamp of newest frame in the kernel (alsa) buffer.
		LOCATION_MAX // for sizing arrays only
		};

		// This needs to be kept in sync with android.media.AudioTimestamp
		enum Timebase {
		TIMEBASE_MONOTONIC, // Clock monotonic offset (generally 0)
		TIMEBASE_BOOTTIME,
		TIMEBASE_MAX,
		};

		ExtendedTimestamp() {
		clear();
		}

		// mPosition is expressed in frame units.
		// It is generally nonnegative, though we keep this signed for
		// to potentially express algorithmic latency at the start of the stream
		// and to prevent unintentional unsigned integer underflow.
		int64_t mPosition[LOCATION_MAX];

		// mTimeNs is in nanoseconds for the default timebase, monotonic.
		// If this value is -1, then both time and position are invalid.
		// If this value is 0, then the time is not valid but the position is valid.
		int64_t mTimeNs[LOCATION_MAX];

		// mTimebaseOffset is the offset in ns from monotonic when the
		// timestamp was taken. This may vary due to suspend time
		// or NTP adjustment.
		int64_t mTimebaseOffset[TIMEBASE_MAX];

		void clear() {
		memset(mPosition, 0, sizeof(mPosition)); // actually not necessary if time is -1
		for (int i = 0; i < LOCATION_MAX; ++i) {
		mTimeNs[i] = -1;
		}
		memset(mTimebaseOffset, 0, sizeof(mTimebaseOffset));
		}

		// Returns the best timestamp as judged from the closest-to-hw stage in the
		// pipeline with a valid timestamp.
		int getBestTimestamp(int64_t position, int64_t time, int timebase) {
		if (position == nullptr \|\| time == nullptr
		\|\| timebase < 0 \|\| timebase >= TIMEBASE_MAX) {
		return BAD_VALUE;
		}
		// look for the closest-to-hw stage in the pipeline with a valid timestamp.
		// We omit LOCATION_CLIENT as we prefer at least LOCATION_SERVER based accuracy
		// when getting the best timestamp.
		for (int i = LOCATION_MAX - 1; i >= LOCATION_SERVER; --i) {
		if (mTimeNs[i] > 0) {
		*position = mPosition[i];
		*time = mTimeNs[i] + mTimebaseOffset[timebase];
		return OK;
		}
		}
		return INVALID_OPERATION;
		}

		// convert fields to a printable string
		std::string toString() {
		std::stringstream ss;

		ss << "BOOTTIME offset " << mTimebaseOffset[TIMEBASE_BOOTTIME] << "\n";
		for (int i = 0; i < LOCATION_MAX; ++i) {
		ss << "ExtendedTimestamp[" << i << "] position: "
		<< mPosition[i] << " time: " << mTimeNs[i] << "\n";
		}
		return ss.str();
		}
		// TODO:
		// Consider adding buffer status:
		// size, available, algorithmic latency
		};

		} // namespace

		#endif // ANDROID_AUDIO_TIMESTAMP_H

include/private/media/AudioTrackShared.h

+51 −3

Original line number	Diff line number	Diff line
		@@ -26,6 +26,7 @@
		#include <utils/RefBase.h>
		#include <audio_utils/roundup.h>
		#include <media/AudioResamplerPublic.h>
		#include <media/AudioTimestamp.h>
		#include <media/Modulo.h>
		#include <media/SingleStateQueue.h>

		@@ -118,6 +119,8 @@ struct AudioTrackSharedStatic {

		typedef SingleStateQueue<AudioPlaybackRate> PlaybackRateQueue;

		typedef SingleStateQueue<ExtendedTimestamp> ExtendedTimestampQueue;

		// ----------------------------------------------------------------------------

		// Important: do not add any virtual methods, including ~
		@@ -171,6 +174,8 @@ private:

		uint16_t mPad2; // unused

		// server write-only, client read
		ExtendedTimestampQueue::Shared mExtendedTimestampQueue;
		public:

		volatile int32_t mFlags; // combinations of CBLK_*
		@@ -426,8 +431,39 @@ public:
		AudioRecordClientProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
		size_t frameSize)
		: ClientProxy(cblk, buffers, frameCount, frameSize,
		false /isOut/, false /clientInServer/) { }
		false /isOut/, false /clientInServer/)
		, mTimestampObserver(&cblk->mExtendedTimestampQueue) { }
		~AudioRecordClientProxy() { }

		status_t getTimestamp(ExtendedTimestamp *timestamp) {
		if (timestamp == nullptr) {
		return BAD_VALUE;
		}
		(void) mTimestampObserver.poll(mTimestamp);
		*timestamp = mTimestamp;
		return OK;
		}

		void clearTimestamp() {
		mTimestamp.clear();
		}

		// Advances the client read pointer to the server write head pointer
		// effectively flushing the client read buffer. The effect is
		// instantaneous. Returns the number of frames flushed.
		uint32_t flush() {
		int32_t rear = android_atomic_acquire_load(&mCblk->u.mStreaming.mRear);
		int32_t front = mCblk->u.mStreaming.mFront;
		android_atomic_release_store(rear, &mCblk->u.mStreaming.mFront);
		return (Modulo<int32_t>(rear) - front).unsignedValue();
		}

		private:
		// The shared buffer contents referred to by the timestamp observer
		// is initialized when the server proxy created. A local zero timestamp
		// is initialized by the client constructor.
		ExtendedTimestampQueue::Observer mTimestampObserver;
		ExtendedTimestamp mTimestamp; // initialized by constructor
		};

		// ----------------------------------------------------------------------------
		@@ -476,6 +512,7 @@ public:
		protected:
		size_t mAvailToClient; // estimated frames available to client prior to releaseBuffer()
		int32_t mFlush; // our copy of cblk->u.mStreaming.mFlush, for streaming output only
		int64_t mReleased; // our copy of cblk->mServer, at 64 bit resolution
		};

		// Proxy used by AudioFlinger for servicing AudioTrack
		@@ -520,7 +557,7 @@ public:
		virtual uint32_t getUnderrunFrames() const { return mCblk->u.mStreaming.mUnderrunFrames; }

		// Return the total number of frames that AudioFlinger has obtained and released
		virtual size_t framesReleased() const { return mCblk->mServer; }
		virtual size_t framesReleased() const { return mReleased; }

		// Return the playback speed and pitch read atomically. Not multi-thread safe on server side.
		AudioPlaybackRate getPlaybackRate();
		@@ -574,9 +611,20 @@ class AudioRecordServerProxy : public ServerProxy {
		public:
		AudioRecordServerProxy(audio_track_cblk_t* cblk, void *buffers, size_t frameCount,
		size_t frameSize, bool clientInServer)
		: ServerProxy(cblk, buffers, frameCount, frameSize, false /isOut/, clientInServer) { }
		: ServerProxy(cblk, buffers, frameCount, frameSize, false /isOut/, clientInServer)
		, mTimestampMutator(&cblk->mExtendedTimestampQueue) { }

		// Return the total number of frames that AudioFlinger has obtained and released
		virtual int64_t framesReleased() const { return mReleased; }

		// Expose timestamp to client proxy. Should only be called by a single thread.
		virtual void setExtendedTimestamp(const ExtendedTimestamp &timestamp) {
		mTimestampMutator.push(timestamp);
		}
		protected:
		virtual ~AudioRecordServerProxy() { }

		ExtendedTimestampQueue::Mutator mTimestampMutator;
		};

		// ----------------------------------------------------------------------------

media/libmedia/AudioRecord.cpp

+41 −1

Original line number	Diff line number	Diff line
		@@ -284,6 +284,8 @@ status_t AudioRecord::set(
		mSequence = 1;
		mObservedSequence = mSequence;
		mInOverrun = false;
		mFramesRead = 0;
		mFramesReadServerOffset = 0;

		return NO_ERROR;
		}
		@@ -299,6 +301,12 @@ status_t AudioRecord::start(AudioSystem::sync_event_t event, int triggerSession)
		return NO_ERROR;
		}

		// discard data in buffer
		const uint32_t framesFlushed = mProxy->flush();
		mFramesReadServerOffset -= mFramesRead + framesFlushed;
		mFramesRead = 0;
		mProxy->clearTimestamp(); // timestamp is invalid until next server push

		// reset current position as seen by client to 0
		mProxy->setEpoch(mProxy->getEpoch() - mProxy->getPosition());
		// force refresh of remaining frames by processAudioBuffer() as last
		@@ -449,6 +457,27 @@ uint32_t AudioRecord::getInputFramesLost() const
		return AudioSystem::getInputFramesLost(getInputPrivate());
		}

		status_t AudioRecord::getTimestamp(ExtendedTimestamp *timestamp)
		{
		if (timestamp == nullptr) {
		return BAD_VALUE;
		}
		AutoMutex lock(mLock);
		status_t status = mProxy->getTimestamp(timestamp);
		if (status == OK) {
		timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesRead;
		timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
		// server side frame offset in case AudioRecord has been restored.
		for (int i = ExtendedTimestamp::LOCATION_SERVER;
		i < ExtendedTimestamp::LOCATION_MAX; ++i) {
		if (timestamp->mTimeNs[i] >= 0) {
		timestamp->mPosition[i] += mFramesReadServerOffset;
		}
		}
		}
		return status;
		}

		// ---- Explicit Routing ---------------------------------------------------
		status_t AudioRecord::setInputDevice(audio_port_handle_t deviceId) {
		AutoMutex lock(mLock);
		@@ -837,7 +866,10 @@ ssize_t AudioRecord::read(void* buffer, size_t userSize, bool blocking)

		releaseBuffer(&audioBuffer);
		}

		if (read > 0) {
		mFramesRead += read / mFrameSize;
		// mFramesReadTime = systemTime(SYSTEM_TIME_MONOTONIC); // not provided at this time.
		}
		return read;
		}

		@@ -988,6 +1020,7 @@ nsecs_t AudioRecord::processAudioBuffer()
		requested = &timeout;
		}

		size_t readFrames = 0;
		while (mRemainingFrames > 0) {

		Buffer audioBuffer;
		@@ -1049,6 +1082,7 @@ nsecs_t AudioRecord::processAudioBuffer()
		}

		releaseBuffer(&audioBuffer);
		readFrames += releasedFrames;

		// FIXME here is where we would repeat EVENT_MORE_DATA again on same advanced buffer
		// if callback doesn't like to accept the full chunk
		@@ -1072,6 +1106,11 @@ nsecs_t AudioRecord::processAudioBuffer()
		#endif

		}
		if (readFrames > 0) {
		AutoMutex lock(mLock);
		mFramesRead += readFrames;
		// mFramesReadTime = systemTime(SYSTEM_TIME_MONOTONIC); // not provided at this time.
		}
		mRemainingFrames = notificationFrames;
		mRetryOnPartialBuffer = true;

		@@ -1096,6 +1135,7 @@ status_t AudioRecord::restoreRecord_l(const char *from)
		// FIXME this fails if we have a new AudioFlinger instance
		result = mAudioRecord->start(AudioSystem::SYNC_EVENT_SAME, 0);
		}
		mFramesReadServerOffset = mFramesRead; // server resets to zero so we need an offset.
		}
		if (result != NO_ERROR) {
		ALOGW("restoreRecord_l() failed status %d", result);

media/libmedia/AudioTrackShared.cpp

+4 −1

Original line number	Diff line number	Diff line
		@@ -597,7 +597,7 @@ void StaticAudioTrackClientProxy::getBufferPositionAndLoopCount(
		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)
		mAvailToClient(0), mFlush(0), mReleased(0)
		{
		}

		@@ -733,6 +733,7 @@ void ServerProxy::releaseBuffer(Buffer* buffer)
		}

		cblk->mServer += stepCount;
		mReleased += stepCount;

		size_t half = mFrameCount / 2;
		if (half == 0) {
		@@ -1033,6 +1034,8 @@ void StaticAudioTrackServerProxy::releaseBuffer(Buffer* buffer)
		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;