am 55a2dcce: Merge "Using centralized isAudioPlaybackRateValid to validate parameters" into mnc-dev (fdb9404f) · Commits · e / os / android_frameworks_av

include/media/AudioResamplerPublic.h

+17 −0

Original line number	Original line	Diff line number	Diff line
	@@ -108,6 +108,23 @@ static inline bool isAudioPlaybackRateEqual(const AudioPlaybackRate &pr1,
	pr2.mFallbackMode == pr2.mFallbackMode;		pr2.mFallbackMode == pr2.mFallbackMode;
	}		}

			static inline bool isAudioPlaybackRateValid(const AudioPlaybackRate &playbackRate) {
			if (playbackRate.mFallbackMode == AUDIO_TIMESTRETCH_FALLBACK_FAIL &&
			(playbackRate.mStretchMode == AUDIO_TIMESTRETCH_STRETCH_SPEECH \|\|
			playbackRate.mStretchMode == AUDIO_TIMESTRETCH_STRETCH_DEFAULT)) {
			//test sonic specific constraints
			return playbackRate.mSpeed >= TIMESTRETCH_SONIC_SPEED_MIN &&
			playbackRate.mSpeed <= TIMESTRETCH_SONIC_SPEED_MAX &&
			playbackRate.mPitch >= AUDIO_TIMESTRETCH_PITCH_MIN &&
			playbackRate.mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX;
			} else {
			return playbackRate.mSpeed >= AUDIO_TIMESTRETCH_SPEED_MIN &&
			playbackRate.mSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX &&
			playbackRate.mPitch >= AUDIO_TIMESTRETCH_PITCH_MIN &&
			playbackRate.mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX;
			}
			}

	// TODO: Consider putting these inlines into a class scope		// TODO: Consider putting these inlines into a class scope

	// Returns the source frames needed to resample to destination frames. This is not a precise		// Returns the source frames needed to resample to destination frames. This is not a precise

media/libmedia/AudioTrack.cpp

+20 −12

Original line number	Original line	Diff line number	Diff line
	@@ -78,7 +78,7 @@ static inline uint32_t adjustSampleRate(uint32_t sampleRate, float pitch)

	static inline float adjustSpeed(float speed, float pitch)		static inline float adjustSpeed(float speed, float pitch)
	{		{
	return kFixPitch ? (speed / pitch) : speed;		return kFixPitch ? speed / max(pitch, AUDIO_TIMESTRETCH_PITCH_MIN_DELTA) : speed;
	}		}

	static inline float adjustPitch(float pitch)		static inline float adjustPitch(float pitch)
	@@ -809,25 +809,33 @@ status_t AudioTrack::setPlaybackRate(const AudioPlaybackRate &playbackRate)
	const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);		const uint32_t effectiveRate = adjustSampleRate(mSampleRate, playbackRate.mPitch);
	const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);		const float effectiveSpeed = adjustSpeed(playbackRate.mSpeed, playbackRate.mPitch);
	const float effectivePitch = adjustPitch(playbackRate.mPitch);		const float effectivePitch = adjustPitch(playbackRate.mPitch);
	if (effectiveSpeed < AUDIO_TIMESTRETCH_SPEED_MIN		AudioPlaybackRate playbackRateTemp = playbackRate;
	\|\| effectiveSpeed > AUDIO_TIMESTRETCH_SPEED_MAX		playbackRateTemp.mSpeed = effectiveSpeed;
	\|\| effectivePitch < AUDIO_TIMESTRETCH_PITCH_MIN		playbackRateTemp.mPitch = effectivePitch;
	\|\| effectivePitch > AUDIO_TIMESTRETCH_PITCH_MAX) {
			if (!isAudioPlaybackRateValid(playbackRateTemp)) {
	return BAD_VALUE;		return BAD_VALUE;
	//TODO: add function in AudioResamplerPublic.h to check for validity.
	}		}
	// Check if the buffer size is compatible.		// Check if the buffer size is compatible.
	if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {		if (!isSampleRateSpeedAllowed_l(effectiveRate, effectiveSpeed)) {
	ALOGV("setPlaybackRate(%f, %f) failed", playbackRate.mSpeed, playbackRate.mPitch);		ALOGV("setPlaybackRate(%f, %f) failed", playbackRate.mSpeed, playbackRate.mPitch);
	return BAD_VALUE;		return BAD_VALUE;
	}		}
	mPlaybackRate = playbackRate;
	mProxy->setPlaybackRate(playbackRate);

	//modify this		// Check resampler ratios are within bounds
	AudioPlaybackRate playbackRateTemp = playbackRate;		if (effectiveRate > mSampleRate * AUDIO_RESAMPLER_DOWN_RATIO_MAX) {
	playbackRateTemp.mSpeed = effectiveSpeed;		ALOGV("setPlaybackRate(%f, %f) failed. Resample rate exceeds max accepted value",
	playbackRateTemp.mPitch = effectivePitch;		playbackRate.mSpeed, playbackRate.mPitch);
			return BAD_VALUE;
			}

			if (effectiveRate * AUDIO_RESAMPLER_UP_RATIO_MAX < mSampleRate) {
			ALOGV("setPlaybackRate(%f, %f) failed. Resample rate below min accepted value",
			playbackRate.mSpeed, playbackRate.mPitch);
			return BAD_VALUE;
			}
			mPlaybackRate = playbackRate;
			//set effective rates
	mProxy->setPlaybackRate(playbackRateTemp);		mProxy->setPlaybackRate(playbackRateTemp);
	mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate		mProxy->setSampleRate(effectiveRate); // FIXME: not quite "atomic" with setPlaybackRate
	return NO_ERROR;		return NO_ERROR;

services/audioflinger/AudioMixer.cpp

+3 −7

Original line number	Original line	Diff line number	Diff line
	@@ -733,13 +733,9 @@ void AudioMixer::setParameter(int name, int target, int param, void *value)
	case PLAYBACK_RATE: {		case PLAYBACK_RATE: {
	const AudioPlaybackRate *playbackRate =		const AudioPlaybackRate *playbackRate =
	reinterpret_cast<AudioPlaybackRate*>(value);		reinterpret_cast<AudioPlaybackRate*>(value);
	ALOG_ASSERT(AUDIO_TIMESTRETCH_SPEED_MIN <= playbackRate->mSpeed		ALOGW_IF(!isAudioPlaybackRateValid(*playbackRate),
	&& playbackRate->mSpeed <= AUDIO_TIMESTRETCH_SPEED_MAX,		"bad parameters speed %f, pitch %f",playbackRate->mSpeed,
	"bad speed %f", playbackRate->mSpeed);		playbackRate->mPitch);
	ALOG_ASSERT(AUDIO_TIMESTRETCH_PITCH_MIN <= playbackRate->mPitch
	&& playbackRate->mPitch <= AUDIO_TIMESTRETCH_PITCH_MAX,
	"bad pitch %f", playbackRate->mPitch);
	//TODO: use function from AudioResamplerPublic.h to test validity.
	if (track.setPlaybackRate(*playbackRate)) {		if (track.setPlaybackRate(*playbackRate)) {
	ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "		ALOGV("setParameter(TIMESTRETCH, PLAYBACK_RATE, STRETCH_MODE, FALLBACK_MODE "
	"%f %f %d %d",		"%f %f %d %d",

services/audioflinger/BufferProviders.cpp

+5 −3

Original line number	Original line	Diff line number	Diff line
	@@ -332,7 +332,8 @@ TimestretchBufferProvider::TimestretchBufferProvider(int32_t channelCount,
	mLocalBufferData(NULL),		mLocalBufferData(NULL),
	mRemaining(0),		mRemaining(0),
	mSonicStream(sonicCreateStream(sampleRate, mChannelCount)),		mSonicStream(sonicCreateStream(sampleRate, mChannelCount)),
	mFallbackFailErrorShown(false)		mFallbackFailErrorShown(false),
			mAudioPlaybackRateValid(false)
	{		{
	LOG_ALWAYS_FATAL_IF(mSonicStream == NULL,		LOG_ALWAYS_FATAL_IF(mSonicStream == NULL,
	"TimestretchBufferProvider can't allocate Sonic stream");		"TimestretchBufferProvider can't allocate Sonic stream");
	@@ -460,6 +461,8 @@ status_t TimestretchBufferProvider::setPlaybackRate(const AudioPlaybackRate &pla
	sonicSetSpeed(mSonicStream, mPlaybackRate.mSpeed);		sonicSetSpeed(mSonicStream, mPlaybackRate.mSpeed);
	//TODO: pitch is ignored for now		//TODO: pitch is ignored for now
	//TODO: optimize: if parameters are the same, don't do any extra computation.		//TODO: optimize: if parameters are the same, don't do any extra computation.

			mAudioPlaybackRateValid = isAudioPlaybackRateValid(mPlaybackRate);
	return OK;		return OK;
	}		}

	@@ -479,8 +482,7 @@ void TimestretchBufferProvider::processFrames(void dstBuffer, size_t dstFrames
	*srcFrames = targetSrc + 1;		*srcFrames = targetSrc + 1;
	}		}

	if (mPlaybackRate.mSpeed< TIMESTRETCH_SONIC_SPEED_MIN \|\|		if (!mAudioPlaybackRateValid) {
	mPlaybackRate.mSpeed > TIMESTRETCH_SONIC_SPEED_MAX ) {
	//fallback mode		//fallback mode
	if (*dstFrames > 0) {		if (*dstFrames > 0) {
	switch(mPlaybackRate.mFallbackMode) {		switch(mPlaybackRate.mFallbackMode) {

services/audioflinger/BufferProviders.h

+1 −0

Original line number	Original line	Diff line number	Diff line
	@@ -188,6 +188,7 @@ private:
	sonicStream mSonicStream; // handle to sonic timestretch object		sonicStream mSonicStream; // handle to sonic timestretch object
	//FIXME: this dependency should be abstracted out		//FIXME: this dependency should be abstracted out
	bool mFallbackFailErrorShown; // log fallback error only once		bool mFallbackFailErrorShown; // log fallback error only once
			bool mAudioPlaybackRateValid; // flag for current parameters validity
	};		};

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