Merge "Implement client playback timestamps with 64 bit accuracy" into nyc-dev (f3bcfebc) · Commits · e / os / android_frameworks_av

include/media/AudioTimestamp.h

+11 −0

Original line number	Diff line number	Diff line
		@@ -69,12 +69,23 @@ struct ExtendedTimestamp {
		// or NTP adjustment.
		int64_t mTimebaseOffset[TIMEBASE_MAX];

		// Playback only:
		// mFlushed is number of flushed frames before entering the server mix;
		// hence not included in mPosition. This is used for adjusting server positions
		// information for frames "dropped".
		// FIXME: This variable should be eliminated, with the offset added on the server side
		// before sending to client, but differences in legacy position offset handling
		// and new extended timestamps require this to be maintained as a separate quantity.
		int64_t mFlushed;

		// Call to reset the timestamp to the original (invalid) state
		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));
		mFlushed = 0;
		}

		// Returns the best timestamp as judged from the closest-to-hw stage in the

include/media/AudioTrack.h

+47 −1

Original line number	Diff line number	Diff line
		@@ -727,12 +727,51 @@ public:
		* because the audio device changed or AudioFlinger died.
		* This typically occurs for direct or offload tracks
		* or if mDoNotReconnect is true.
		* INVALID_OPERATION if called on a FastTrack, wrong state, or some other error.
		* INVALID_OPERATION wrong state, or some other error.
		*
		* The timestamp parameter is undefined on return, if status is not NO_ERROR.
		*/
		status_t getTimestamp(AudioTimestamp& timestamp);

		/* Return the extended timestamp, with additional timebase info and improved drain behavior.
		*
		* This is similar to the AudioTrack.java API:
		* getTimestamp(@NonNull AudioTimestamp timestamp, @AudioTimestamp.Timebase int timebase)
		*
		* Some differences between this method and the getTimestamp(AudioTimestamp& timestamp) method
		*
		* 1. stop() by itself does not reset the frame position.
		* A following start() resets the frame position to 0.
		* 2. flush() by itself does not reset the frame position.
		* The frame position advances by the number of frames flushed,
		* when the first frame after flush reaches the audio sink.
		* 3. BOOTTIME clock offsets are provided to help synchronize with
		* non-audio streams, e.g. sensor data.
		* 4. Position is returned with 64 bits of resolution.
		*
		* Parameters:
		* timestamp: A pointer to the caller allocated ExtendedTimestamp.
		*
		* Returns NO_ERROR on success; timestamp is filled with valid data.
		* BAD_VALUE if timestamp is NULL.
		* WOULD_BLOCK if called immediately after start() when the number
		* of frames consumed is less than the
		* overall hardware latency to physical output. In WOULD_BLOCK cases,
		* one might poll again, or use getPosition(), or use 0 position and
		* current time for the timestamp.
		* If WOULD_BLOCK is returned, the timestamp is still
		* modified with the LOCATION_CLIENT portion filled.
		* DEAD_OBJECT if AudioFlinger dies or the output device changes and
		* the track cannot be automatically restored.
		* The application needs to recreate the AudioTrack
		* because the audio device changed or AudioFlinger died.
		* This typically occurs for direct or offloaded tracks
		* or if mDoNotReconnect is true.
		* INVALID_OPERATION if called on a offloaded or direct track.
		* Use getTimestamp(AudioTimestamp& timestamp) instead.
		*/
		status_t getTimestamp(ExtendedTimestamp *timestamp);

		/* Add an AudioDeviceCallback. The caller will be notified when the audio device to which this
		* AudioTrack is routed is updated.
		* Replaces any previously installed callback.
		@@ -956,6 +995,13 @@ protected:

		uint32_t mUnderrunCountOffset; // updated when restoring tracks

		int64_t mFramesWritten; // total frames written. reset to zero after
		// the start() following stop(). It is not
		// changed after restoring the track or
		// after flush.
		int64_t mFramesWrittenServerOffset; // An offset to server frames due to
		// restoring AudioTrack, or stop/start.

		audio_output_flags_t mFlags; // same as mOrigFlags, except for bits that may
		// be denied by client or server, such as
		// AUDIO_OUTPUT_FLAG_FAST. mLock must be

include/private/media/AudioTrackShared.h

+4 −1

Original line number	Diff line number	Diff line
		@@ -522,6 +522,9 @@ public:
		mTimestampMutator.push(timestamp);
		}

		// Total count of the number of flushed frames since creation (never reset).
		virtual int64_t framesFlushed() const { return mFlushed; }

		// Get dynamic buffer size from the shared control block.
		uint32_t getBufferSizeInFrames() const {
		return android_atomic_acquire_load((int32_t *)&mCblk->mBufferSizeInFrames);
		@@ -531,7 +534,7 @@ 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

		int64_t mFlushed; // flushed frames to account for client-server discrepancy
		ExtendedTimestampQueue::Mutator mTimestampMutator;
		};

media/libmedia/AudioTrack.cpp

+56 −0

Original line number	Diff line number	Diff line
		@@ -500,6 +500,8 @@ status_t AudioTrack::set(
		mTimestampStartupGlitchReported = false;
		mRetrogradeMotionReported = false;
		mUnderrunCountOffset = 0;
		mFramesWritten = 0;
		mFramesWrittenServerOffset = 0;

		return NO_ERROR;
		}
		@@ -537,6 +539,14 @@ status_t AudioTrack::start()
		// Note: the if is technically unnecessary because previousState == STATE_FLUSHED
		// is only for streaming tracks, and mMarkerReached is already set to false.
		if (previousState == STATE_STOPPED) {
		// read last server side position change via timestamp
		ExtendedTimestamp ets;
		if (mProxy->getTimestamp(&ets) == OK &&
		ets.mTimeNs[ExtendedTimestamp::LOCATION_SERVER] > 0) {
		mFramesWrittenServerOffset = -(ets.mPosition[ExtendedTimestamp::LOCATION_SERVER]
		+ ets.mFlushed);
		}
		mFramesWritten = 0;
		mProxy->clearTimestamp(); // need new server push for valid timestamp
		mMarkerReached = false;
		}
		@@ -1657,6 +1667,9 @@ ssize_t AudioTrack::write(const void* buffer, size_t userSize, bool blocking)
		releaseBuffer(&audioBuffer);
		}

		if (written > 0) {
		mFramesWritten += written / mFrameSize;
		}
		return written;
		}

		@@ -1923,6 +1936,7 @@ nsecs_t AudioTrack::processAudioBuffer()
		requested = &timeout;
		}

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

		Buffer audioBuffer;
		@@ -2024,6 +2038,7 @@ nsecs_t AudioTrack::processAudioBuffer()
		}

		releaseBuffer(&audioBuffer);
		writtenFrames += 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
		@@ -2047,6 +2062,10 @@ nsecs_t AudioTrack::processAudioBuffer()
		#endif

		}
		if (writtenFrames > 0) {
		AutoMutex lock(mLock);
		mFramesWritten += writtenFrames;
		}
		mRemainingFrames = notificationFrames;
		mRetryOnPartialBuffer = true;

		@@ -2109,6 +2128,7 @@ status_t AudioTrack::restoreTrack_l(const char *from)
		}
		if (mState == STATE_ACTIVE) {
		result = mAudioTrack->start();
		mFramesWrittenServerOffset = mFramesWritten; // server resets to zero so we offset
		}
		}
		if (result != NO_ERROR) {
		@@ -2162,6 +2182,42 @@ status_t AudioTrack::setParameters(const String8& keyValuePairs)
		return mAudioTrack->setParameters(keyValuePairs);
		}

		status_t AudioTrack::getTimestamp(ExtendedTimestamp *timestamp)
		{
		if (timestamp == nullptr) {
		return BAD_VALUE;
		}
		AutoMutex lock(mLock);
		if (mCblk->mFlags & CBLK_INVALID) {
		const status_t status = restoreTrack_l("getTimestampExtended");
		if (status != OK) {
		// per getTimestamp() API doc in header, we return DEAD_OBJECT here,
		// recommending that the track be recreated.
		return DEAD_OBJECT;
		}
		}
		// check for offloaded/direct here in case restoring somehow changed those flags.
		if (isOffloadedOrDirect_l()) {
		return INVALID_OPERATION; // not supported
		}
		status_t status = mProxy->getTimestamp(timestamp);
		bool found = false;
		if (status == OK) {
		timestamp->mPosition[ExtendedTimestamp::LOCATION_CLIENT] = mFramesWritten;
		timestamp->mTimeNs[ExtendedTimestamp::LOCATION_CLIENT] = 0;
		// server side frame offset in case AudioTrack has been restored.
		for (int i = ExtendedTimestamp::LOCATION_SERVER;
		i < ExtendedTimestamp::LOCATION_MAX; ++i) {
		if (timestamp->mTimeNs[i] >= 0) {
		// apply server offset and the "flush frame correction here"
		timestamp->mPosition[i] += mFramesWrittenServerOffset + timestamp->mFlushed;
		found = true;
		}
		}
		}
		return found ? OK : WOULD_BLOCK;
		}

		status_t AudioTrack::getTimestamp(AudioTimestamp& timestamp)
		{
		AutoMutex lock(mLock);

media/libmedia/AudioTrackShared.cpp

+2 −1

Original line number	Diff line number	Diff line
		@@ -615,7 +615,7 @@ 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), mReleased(0)
		mAvailToClient(0), mFlush(0), mReleased(0), mFlushed(0)
		, mTimestampMutator(&cblk->mExtendedTimestampQueue)
		{
		cblk->mBufferSizeInFrames = frameCount;
		@@ -671,6 +671,7 @@ status_t ServerProxy::obtainBuffer(Buffer* buffer, bool ackFlush)
		mClientInServer ? FUTEX_WAKE_PRIVATE : FUTEX_WAKE, 1);
		}
		}
		mFlushed += (newFront - front) & mask;
		front = newFront;
		}
		} else {