Merge "NuPlayer: Reserve buffer for 8x max speed" into nyc-dev (0c4a97e6) · Commits · e / os / android_frameworks_av

include/media/AudioTrack.h

+11 −3

Original line number	Diff line number	Diff line
		@@ -201,6 +201,10 @@ public:
		binder to AudioFlinger.
		It will return an error instead. The application will recreate
		the track based on offloading or different channel configuration, etc.
		* maxRequiredSpeed: For PCM tracks, this creates an appropriate buffer size that will allow
		* maxRequiredSpeed playback. Values less than 1.0f and greater than
		* AUDIO_TIMESTRETCH_SPEED_MAX will be clamped. For non-PCM tracks
		* and direct or offloaded tracks, this parameter is ignored.
		* threadCanCallJava: Not present in parameter list, and so is fixed at false.
		*/

		@@ -219,7 +223,8 @@ public:
		int uid = -1,
		pid_t pid = -1,
		const audio_attributes_t* pAttributes = NULL,
		bool doNotReconnect = false);
		bool doNotReconnect = false,
		float maxRequiredSpeed = 1.0f);

		/* Creates an audio track and registers it with AudioFlinger.
		* With this constructor, the track is configured for static buffer mode.
		@@ -248,7 +253,8 @@ public:
		int uid = -1,
		pid_t pid = -1,
		const audio_attributes_t* pAttributes = NULL,
		bool doNotReconnect = false);
		bool doNotReconnect = false,
		float maxRequiredSpeed = 1.0f);

		/* Terminates the AudioTrack and unregisters it from AudioFlinger.
		* Also destroys all resources associated with the AudioTrack.
		@@ -293,7 +299,8 @@ public:
		int uid = -1,
		pid_t pid = -1,
		const audio_attributes_t* pAttributes = NULL,
		bool doNotReconnect = false);
		bool doNotReconnect = false,
		float maxRequiredSpeed = 1.0f);

		/* Result of constructing the AudioTrack. This must be checked for successful initialization
		* before using any AudioTrack API (except for set()), because using
		@@ -920,6 +927,7 @@ protected:
		mutable uint32_t mSampleRate; // mutable because getSampleRate() can update it
		uint32_t mOriginalSampleRate;
		AudioPlaybackRate mPlaybackRate;
		float mMaxRequiredSpeed; // use PCM buffer size to allow this speed

		// Corresponds to current IAudioTrack, value is reported back by AudioFlinger to the client.
		// This allocated buffer size is maintained by the proxy.

media/libmedia/AudioTrack.cpp

+23 −7

Original line number	Diff line number	Diff line
		@@ -191,7 +191,8 @@ AudioTrack::AudioTrack(
		int uid,
		pid_t pid,
		const audio_attributes_t* pAttributes,
		bool doNotReconnect)
		bool doNotReconnect,
		float maxRequiredSpeed)
		: mStatus(NO_INIT),
		mState(STATE_STOPPED),
		mPreviousPriority(ANDROID_PRIORITY_NORMAL),
		@@ -202,7 +203,7 @@ AudioTrack::AudioTrack(
		mStatus = set(streamType, sampleRate, format, channelMask,
		frameCount, flags, cbf, user, notificationFrames,
		0 /sharedBuffer/, false /threadCanCallJava/, sessionId, transferType,
		offloadInfo, uid, pid, pAttributes, doNotReconnect);
		offloadInfo, uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
		}

		AudioTrack::AudioTrack(
		@@ -221,7 +222,8 @@ AudioTrack::AudioTrack(
		int uid,
		pid_t pid,
		const audio_attributes_t* pAttributes,
		bool doNotReconnect)
		bool doNotReconnect,
		float maxRequiredSpeed)
		: mStatus(NO_INIT),
		mState(STATE_STOPPED),
		mPreviousPriority(ANDROID_PRIORITY_NORMAL),
		@@ -232,7 +234,7 @@ AudioTrack::AudioTrack(
		mStatus = set(streamType, sampleRate, format, channelMask,
		0 /frameCount/, flags, cbf, user, notificationFrames,
		sharedBuffer, false /threadCanCallJava/, sessionId, transferType, offloadInfo,
		uid, pid, pAttributes, doNotReconnect);
		uid, pid, pAttributes, doNotReconnect, maxRequiredSpeed);
		}

		AudioTrack::~AudioTrack()
		@@ -281,7 +283,8 @@ status_t AudioTrack::set(
		int uid,
		pid_t pid,
		const audio_attributes_t* pAttributes,
		bool doNotReconnect)
		bool doNotReconnect,
		float maxRequiredSpeed)
		{
		ALOGV("set(): streamType %d, sampleRate %u, format %#x, channelMask %#x, frameCount %zu, "
		"flags #%x, notificationFrames %u, sessionId %d, transferType %d, uid %d, pid %d",
		@@ -422,6 +425,8 @@ status_t AudioTrack::set(
		mSampleRate = sampleRate;
		mOriginalSampleRate = sampleRate;
		mPlaybackRate = AUDIO_PLAYBACK_RATE_DEFAULT;
		// 1.0 <= mMaxRequiredSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX
		mMaxRequiredSpeed = min(max(maxRequiredSpeed, 1.0f), AUDIO_TIMESTRETCH_SPEED_MAX);

		// Make copy of input parameter offloadInfo so that in the future:
		// (a) createTrack_l doesn't need it as an input parameter
		@@ -824,6 +829,9 @@ status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
		if (mFlags & AUDIO_OUTPUT_FLAG_FAST) {
		return INVALID_OPERATION;
		}

		ALOGV("setPlaybackRate (input): mSampleRate:%u mSpeed:%f mPitch:%f",
		mSampleRate, playbackRate.mSpeed, playbackRate.mPitch);
		// pitch is emulated by adjusting speed and sampleRate
		const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
		const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
		@@ -832,12 +840,18 @@ status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
		playbackRateTemp.mSpeed = effectiveSpeed;
		playbackRateTemp.mPitch = effectivePitch;

		ALOGV("setPlaybackRate (effective): mSampleRate:%u mSpeed:%f mPitch:%f",
		effectiveRate, effectiveSpeed, effectivePitch);

		if (!isAudioPlaybackRateValid(playbackRateTemp)) {
		ALOGV("setPlaybackRate(%f, %f) failed (effective rate out of bounds)",
		playbackRate.mSpeed, playbackRate.mPitch);
		return BAD_VALUE;
		}
		// Check if the buffer size is compatible.
		if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
		ALOGV("setPlaybackRate(%f, %f) failed", playbackRate.mSpeed, playbackRate.mPitch);
		ALOGV("setPlaybackRate(%f, %f) failed (buffer size)",
		playbackRate.mSpeed, playbackRate.mPitch);
		return BAD_VALUE;
		}

		@@ -1293,9 +1307,11 @@ status_t AudioTrack::createTrack_l()

		if ((mFlags & AUDIO_OUTPUT_FLAG_FAST) == 0) {
		// for normal tracks precompute the frame count based on speed.
		const float speed = !isPurePcmData_l() \|\| isOffloadedOrDirect_l() ? 1.0f :
		max(mMaxRequiredSpeed, mPlaybackRate.mSpeed);
		const size_t minFrameCount = calculateMinFrameCount(
		mAfLatency, mAfFrameCount, mAfSampleRate, mSampleRate,
		mPlaybackRate.mSpeed);
		speed);
		if (frameCount < minFrameCount) {
		frameCount = minFrameCount;
		}

media/libmediaplayerservice/MediaPlayerService.cpp

+12 −1

Original line number	Diff line number	Diff line
		@@ -94,6 +94,8 @@ using android::Parcel;
		// Max number of entries in the filter.
		const int kMaxFilterSize = 64; // I pulled that out of thin air.

		const float kMaxRequiredSpeed = 8.0f; // for PCM tracks allow up to 8x speedup.

		// FIXME: Move all the metadata related function in the Metadata.cpp


		@@ -1749,6 +1751,14 @@ status_t MediaPlayerService::AudioOutput::open(
		mAttributes,
		doNotReconnect);
		} else {
		// TODO: Due to buffer memory concerns, we use a max target playback speed
		// based on mPlaybackRate at the time of open (instead of kMaxRequiredSpeed),
		// also clamping the target speed to 1.0 <= targetSpeed <= kMaxRequiredSpeed.
		const float targetSpeed =
		std::min(std::max(mPlaybackRate.mSpeed, 1.0f), kMaxRequiredSpeed);
		ALOGW_IF(targetSpeed != mPlaybackRate.mSpeed,
		"track target speed:%f clamped from playback speed:%f",
		targetSpeed, mPlaybackRate.mSpeed);
		t = new AudioTrack(
		mStreamType,
		sampleRate,
		@@ -1765,7 +1775,8 @@ status_t MediaPlayerService::AudioOutput::open(
		mUid,
		mPid,
		mAttributes,
		doNotReconnect);
		doNotReconnect,
		targetSpeed);
		}

		if ((t == 0) \|\| (t->initCheck() != NO_ERROR)) {

media/libmediaplayerservice/nuplayer/NuPlayerRenderer.cpp

+4 −3

Original line number	Diff line number	Diff line
		@@ -1883,6 +1883,10 @@ status_t NuPlayer::Renderer::onOpenAudioSink(
		// NuPlayer a chance to switch from non-offload mode to offload mode.
		// So we only set doNotReconnect when there's no video.
		const bool doNotReconnect = !hasVideo;

		// We should always be able to set our playback settings if the sink is closed.
		LOG_ALWAYS_FATAL_IF(mAudioSink->setPlaybackRate(mPlaybackSettings) != OK,
		"onOpenAudioSink: can't set playback rate on closed sink");
		status_t err = mAudioSink->open(
		sampleRate,
		numChannels,
		@@ -1895,9 +1899,6 @@ status_t NuPlayer::Renderer::onOpenAudioSink(
		NULL,
		doNotReconnect,
		frameCount);
		if (err == OK) {
		err = mAudioSink->setPlaybackRate(mPlaybackSettings);
		}
		if (err != OK) {
		ALOGW("openAudioSink: non offloaded open failed status: %d", err);
		mAudioSink->close();