Snap for 4801384 from dddee5810b848bc7090e7e687cbedcd7beafeb18 to pi-release (de7bfb76) · Commits · e / os / android_frameworks_av

drm/libmediadrm/CryptoHal.cpp

+11 −0

Original line number	Diff line number	Diff line
		@@ -262,6 +262,17 @@ int32_t CryptoHal::setHeapBase(const sp<IMemoryHeap>& heap) {
		void CryptoHal::clearHeapBase(int32_t seqNum) {
		Mutex::Autolock autoLock(mLock);

		/*
		* Clear the remote shared memory mapping by setting the shared
		* buffer base to a null hidl_memory.
		*
		* TODO: Add a releaseSharedBuffer method in a future DRM HAL
		* API version to make this explicit.
		*/
		uint32_t bufferId = mHeapBases.valueFor(seqNum).getBufferId();
		Return<void> hResult = mPlugin->setSharedBufferBase(hidl_memory(), bufferId);
		ALOGE_IF(!hResult.isOk(), "setSharedBufferBase(): remote call failed");

		mHeapBases.removeItem(seqNum);
		}

media/libaudioprocessing/AudioMixer.cpp

+1 −2

Original line number	Diff line number	Diff line
		@@ -192,7 +192,6 @@ bool AudioMixer::setChannelMasks(int name,
		// always recompute for both channel masks even if only one has changed.
		const uint32_t trackChannelCount = audio_channel_count_from_out_mask(trackChannelMask);
		const uint32_t mixerChannelCount = audio_channel_count_from_out_mask(mixerChannelMask);
		const bool mixerChannelCountChanged = track->mMixerChannelCount != mixerChannelCount;

		ALOG_ASSERT((trackChannelCount <= MAX_NUM_CHANNELS_TO_DOWNMIX)
		&& trackChannelCount
		@@ -213,7 +212,7 @@ bool AudioMixer::setChannelMasks(int name,
		// do it after downmix since track format may change!
		track->prepareForReformat();

		if (track->mResampler.get() != nullptr && mixerChannelCountChanged) {
		if (track->mResampler.get() != nullptr) {
		// resampler channels may have changed.
		const uint32_t resetToSampleRate = track->sampleRate;
		track->mResampler.reset(nullptr);

media/libeffects/dynamicsproc/dsp/DPFrequency.cpp

+39 −37

Original line number	Diff line number	Diff line
		@@ -533,41 +533,45 @@ size_t DPFrequency::processFirstStages(ChannelBuffer &cb) {
		fTheta = exp(-1.0 / (fFRelSec * mBlocksPerSecond));
		}


		float fEnv = (1.0 - fTheta) * fEnergySum + fTheta * pMbcBandParams->previousEnvelope;
		//preserve for next iteration
		pMbcBandParams->previousEnvelope = fEnv;

		float fThreshold = dBtoLinear(pMbcBandParams->thresholdDb);
		float fNoiseGateThreshold = dBtoLinear(pMbcBandParams->noiseGateThresholdDb);

		float fNewFactor = 1.0;

		if (fEnv > fThreshold) {
		float fDbAbove = linearToDb(fThreshold / fEnv);
		float fDbTarget = fDbAbove / pMbcBandParams->ratio;
		float fDbChange = fDbAbove - fDbTarget;
		fNewFactor = dBtoLinear(fDbChange);
		} else if (fEnv < fNoiseGateThreshold) {
		if (fEnv < MIN_ENVELOPE) {
		fEnv = MIN_ENVELOPE;
		}
		float fDbBelow = linearToDb(fNoiseGateThreshold / fEnv);
		float fDbTarget = fDbBelow / pMbcBandParams->expanderRatio;
		float fDbChange = fDbBelow - fDbTarget;
		fNewFactor = dBtoLinear(fDbChange);
		}
		const float envDb = linearToDb(fEnv);
		float newLevelDb = envDb;
		//using shorter variables for code clarity
		const float thresholdDb = pMbcBandParams->thresholdDb;
		const float ratio = pMbcBandParams->ratio;
		const float kneeWidthDbHalf = pMbcBandParams->kneeWidthDb / 2;
		const float noiseGateThresholdDb = pMbcBandParams->noiseGateThresholdDb;
		const float expanderRatio = pMbcBandParams->expanderRatio;

		//find segment
		if (envDb > thresholdDb + kneeWidthDbHalf) {
		//compression segment
		newLevelDb = envDb + ((1 / ratio) - 1) * (envDb - thresholdDb);
		} else if (envDb > thresholdDb - kneeWidthDbHalf) {
		//knee-compression segment
		float temp = (envDb - thresholdDb + kneeWidthDbHalf);
		newLevelDb = envDb + ((1 / ratio) - 1) *
		temp * temp / (kneeWidthDbHalf * 4);
		} else if (envDb < noiseGateThresholdDb) {
		//expander segment
		newLevelDb = noiseGateThresholdDb -
		expanderRatio * (noiseGateThresholdDb - envDb);
		}

		float newFactor = dBtoLinear(newLevelDb - envDb);

		//apply post gain.
		fNewFactor *= dBtoLinear(pMbcBandParams->gainPostDb);

		if (fNewFactor < 0) {
		fNewFactor = 0;
		}
		newFactor *= dBtoLinear(pMbcBandParams->gainPostDb);

		//apply to this band
		for (size_t k = pMbcBandParams->binStart; k <= pMbcBandParams->binStop; k++) {
		cb.complexTemp[k] *= fNewFactor;
		cb.complexTemp[k] *= newFactor;
		}

		} //end per band process
		@@ -604,22 +608,20 @@ size_t DPFrequency::processFirstStages(ChannelBuffer &cb) {
		//preserve for next iteration
		cb.mLimiterParams.previousEnvelope = fEnv;

		float fThreshold = dBtoLinear(cb.mLimiterParams.thresholdDb);
		const float envDb = linearToDb(fEnv);
		float newFactorDb = 0;
		//using shorter variables for code clarity
		const float thresholdDb = cb.mLimiterParams.thresholdDb;
		const float ratio = cb.mLimiterParams.ratio;

		float fNewFactor = 1.0;

		if (fEnv > fThreshold) {
		float fDbAbove = linearToDb(fThreshold / fEnv);
		float fDbTarget = fDbAbove / cb.mLimiterParams.ratio;
		float fDbChange = fDbAbove - fDbTarget;
		fNewFactor = dBtoLinear(fDbChange);
		if (envDb > thresholdDb) {
		//limiter segment
		newFactorDb = ((1 / ratio) - 1) * (envDb - thresholdDb);
		}

		if (fNewFactor < 0) {
		fNewFactor = 0;
		}
		float newFactor = dBtoLinear(newFactorDb);

		cb.mLimiterParams.newFactor = fNewFactor;
		cb.mLimiterParams.newFactor = newFactor;

		} //end Limiter
		return mBlockSize;

media/libstagefright/foundation/AMessage.cpp

+1 −0

Original line number	Diff line number	Diff line
		@@ -197,6 +197,7 @@ AMessage::Item AMessage::allocateItem(const char name) {
		CHECK(mNumItems < kMaxNumItems);
		i = mNumItems++;
		item = &mItems[i];
		item->mType = kTypeInt32;
		item->setName(name, len);
		}

services/audiopolicy/common/managerdefinitions/include/SessionRoute.h

+3 −1

Original line number	Diff line number	Diff line
		@@ -24,6 +24,7 @@
		namespace android {

		class DeviceDescriptor;
		class DeviceVector;

		class SessionRoute : public RefBase
		{
		@@ -98,7 +99,8 @@ public:
		int decRouteActivity(audio_session_t session);
		bool getAndClearRouteChanged(audio_session_t session); // also clears the changed flag
		void log(const char* caption);

		audio_devices_t getActiveDeviceForStream(audio_stream_type_t streamType,
		const DeviceVector& availableDevices);
		// Specify an Output(Sink) route by passing SessionRoute::SOURCE_TYPE_NA in the
		// source argument.
		// Specify an Input(Source) rout by passing SessionRoute::AUDIO_STREAM_DEFAULT