Merge "AudioEffect: Convert aux buffer format from q4.27 to float" (d4664356) · Commits · e / os / android_frameworks_av

media/libaudioclient/include/media/AudioMixer.h

+3 −0

Original line number	Diff line number	Diff line
		@@ -33,6 +33,9 @@
		// FIXME This is actually unity gain, which might not be max in future, expressed in U.12
		#define MAX_GAIN_INT AudioMixer::UNITY_GAIN_INT

		// This must match frameworks/av/services/audioflinger/Configuration.h
		#define FLOAT_AUX

		namespace android {

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

media/libaudioprocessing/AudioMixer.cpp

+67 −38

Original line number	Diff line number	Diff line
		@@ -71,12 +71,20 @@ T max(const T& x, const T& y) {

		// Set kUseNewMixer to true to use the new mixer engine always. Otherwise the
		// original code will be used for stereo sinks, the new mixer for multichannel.
		static const bool kUseNewMixer = true;
		static constexpr bool kUseNewMixer = true;

		// Set kUseFloat to true to allow floating input into the mixer engine.
		// If kUseNewMixer is false, this is ignored or may be overridden internally
		// because of downmix/upmix support.
		static const bool kUseFloat = true;
		static constexpr bool kUseFloat = true;

		#ifdef FLOAT_AUX
		using TYPE_AUX = float;
		static_assert(kUseNewMixer && kUseFloat,
		"kUseNewMixer and kUseFloat must be true for FLOAT_AUX option");
		#else
		using TYPE_AUX = int32_t; // q4.27
		#endif

		// Set to default copy buffer size in frames for input processing.
		static const size_t kCopyBufferFrameCount = 256;
		@@ -861,16 +869,25 @@ inline void AudioMixer::track_t::adjustVolumeRamp(bool aux, bool useFloat)
		}
		}
		}
		/* TODO: aux is always integer regardless of output buffer type */

		if (aux) {
		if (((auxInc>0) && (((prevAuxLevel+auxInc)>>16) >= auxLevel)) \|\|
		((auxInc<0) && (((prevAuxLevel+auxInc)>>16) <= auxLevel))) {
		#ifdef FLOAT_AUX
		if (useFloat) {
		if ((mAuxInc > 0.f && mPrevAuxLevel + mAuxInc >= mAuxLevel) \|\|
		(mAuxInc < 0.f && mPrevAuxLevel + mAuxInc <= mAuxLevel)) {
		auxInc = 0;
		prevAuxLevel = auxLevel << 16;
		mAuxInc = 0.;
		mAuxInc = 0.f;
		mPrevAuxLevel = mAuxLevel;
		}
		} else
		#endif
		if ((auxInc > 0 && ((prevAuxLevel + auxInc) >> 16) >= auxLevel) \|\|
		(auxInc < 0 && ((prevAuxLevel + auxInc) >> 16) <= auxLevel)) {
		auxInc = 0;
		prevAuxLevel = auxLevel << 16;
		mAuxInc = 0.f;
		mPrevAuxLevel = mAuxLevel;
		} else {
		//ALOGV("aux ramp: %d %d %d", auxLevel << 16, prevAuxLevel, auxInc);
		}
		}
		}
		@@ -1694,7 +1711,7 @@ void AudioMixer::process__OneTrack16BitsStereoNoResampling(state_t* state)
		/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
		* TO: int32_t (Q4.27) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TA: int32_t (Q4.27)
		* TA: int32_t (Q4.27) or float
		*/
		template <int MIXTYPE,
		typename TO, typename TI, typename TV, typename TA, typename TAV>
		@@ -1738,7 +1755,7 @@ static void volumeRampMulti(uint32_t channels, TO* out, size_t frameCount,
		/* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
		* TO: int32_t (Q4.27) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TA: int32_t (Q4.27)
		* TA: int32_t (Q4.27) or float
		*/
		template <int MIXTYPE,
		typename TO, typename TI, typename TV, typename TA, typename TAV>
		@@ -1778,7 +1795,7 @@ static void volumeMulti(uint32_t channels, TO* out, size_t frameCount,
		* ADJUSTVOL (set to true if volume ramp parameters needs adjustment afterwards)
		* TO: int32_t (Q4.27) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TA: int32_t (Q4.27)
		* TA: int32_t (Q4.27) or float
		*/
		template <int MIXTYPE, bool USEFLOATVOL, bool ADJUSTVOL,
		typename TO, typename TI, typename TA>
		@@ -1788,13 +1805,25 @@ void AudioMixer::volumeMix(TO *out, size_t outFrames,
		if (USEFLOATVOL) {
		if (ramp) {
		volumeRampMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
		t->mPrevVolume, t->mVolumeInc, &t->prevAuxLevel, t->auxInc);
		t->mPrevVolume, t->mVolumeInc,
		#ifdef FLOAT_AUX
		&t->mPrevAuxLevel, t->mAuxInc
		#else
		&t->prevAuxLevel, t->auxInc
		#endif
		);
		if (ADJUSTVOL) {
		t->adjustVolumeRamp(aux != NULL, true);
		}
		} else {
		volumeMulti<MIXTYPE>(t->mMixerChannelCount, out, outFrames, in, aux,
		t->mVolume, t->auxLevel);
		t->mVolume,
		#ifdef FLOAT_AUX
		t->mAuxLevel
		#else
		t->auxLevel
		#endif
		);
		}
		} else {
		if (ramp) {
		@@ -1851,7 +1880,7 @@ void AudioMixer::process_NoResampleOneTrack(state_t* state)
		}

		const size_t outFrames = b.frameCount;
		volumeMix<MIXTYPE, is_same<TI, float>::value, false> (
		volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL /, false / ADJUSTVOL */> (
		out, outFrames, in, aux, ramp, t);

		out += outFrames * channels;
		@@ -1874,7 +1903,7 @@ void AudioMixer::process_NoResampleOneTrack(state_t* state)
		* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
		* TO: int32_t (Q4.27) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TA: int32_t (Q4.27)
		* TA: int32_t (Q4.27) or float
		*/
		template <int MIXTYPE, typename TO, typename TI, typename TA>
		void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO* temp, TA* aux)
		@@ -1890,7 +1919,7 @@ void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO*
		memset(temp, 0, outFrameCount * t->mMixerChannelCount * sizeof(TO));
		t->resampler->resample((int32_t*)temp, outFrameCount, t->bufferProvider);

		volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
		volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL /, true / ADJUSTVOL */>(
		out, outFrameCount, temp, aux, ramp, t);

		} else { // constant volume gain
		@@ -1905,7 +1934,7 @@ void AudioMixer::track__Resample(track_t* t, TO* out, size_t outFrameCount, TO*
		* MIXTYPE (see AudioMixerOps.h MIXTYPE_* enumeration)
		* TO: int32_t (Q4.27) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TA: int32_t (Q4.27)
		* TA: int32_t (Q4.27) or float
		*/
		template <int MIXTYPE, typename TO, typename TI, typename TA>
		void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
		@@ -1914,7 +1943,7 @@ void AudioMixer::track__NoResample(track_t* t, TO* out, size_t frameCount,
		ALOGVV("track__NoResample\n");
		const TI in = static_cast<const TI >(t->in);

		volumeMix<MIXTYPE, is_same<TI, float>::value, true>(
		volumeMix<MIXTYPE, is_same<TI, float>::value /* USEFLOATVOL /, true / ADJUSTVOL */>(
		out, frameCount, in, aux, t->needsRamp(), t);

		// MIXTYPE_MONOEXPAND reads a single input channel and expands to NCHAN output channels.
		@@ -1990,11 +2019,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount
		case TRACKTYPE_RESAMPLE:
		switch (mixerInFormat) {
		case AUDIO_FORMAT_PCM_FLOAT:
		return (AudioMixer::hook_t)
		track__Resample<MIXTYPE_MULTI, float /TO/, float /TI/, int32_t /TA/>;
		return (AudioMixer::hook_t)track__Resample<
		MIXTYPE_MULTI, float /TO/, float /TI/, TYPE_AUX>;
		case AUDIO_FORMAT_PCM_16_BIT:
		return (AudioMixer::hook_t)\
		track__Resample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
		return (AudioMixer::hook_t)track__Resample<
		MIXTYPE_MULTI, int32_t /TO/, int16_t /TI/, TYPE_AUX>;
		default:
		LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
		break;
		@@ -2003,11 +2032,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount
		case TRACKTYPE_NORESAMPLEMONO:
		switch (mixerInFormat) {
		case AUDIO_FORMAT_PCM_FLOAT:
		return (AudioMixer::hook_t)
		track__NoResample<MIXTYPE_MONOEXPAND, float, float, int32_t>;
		return (AudioMixer::hook_t)track__NoResample<
		MIXTYPE_MONOEXPAND, float /TO/, float /TI/, TYPE_AUX>;
		case AUDIO_FORMAT_PCM_16_BIT:
		return (AudioMixer::hook_t)
		track__NoResample<MIXTYPE_MONOEXPAND, int32_t, int16_t, int32_t>;
		return (AudioMixer::hook_t)track__NoResample<
		MIXTYPE_MONOEXPAND, int32_t /TO/, int16_t /TI/, TYPE_AUX>;
		default:
		LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
		break;
		@@ -2016,11 +2045,11 @@ AudioMixer::hook_t AudioMixer::getTrackHook(int trackType, uint32_t channelCount
		case TRACKTYPE_NORESAMPLE:
		switch (mixerInFormat) {
		case AUDIO_FORMAT_PCM_FLOAT:
		return (AudioMixer::hook_t)
		track__NoResample<MIXTYPE_MULTI, float, float, int32_t>;
		return (AudioMixer::hook_t)track__NoResample<
		MIXTYPE_MULTI, float /TO/, float /TI/, TYPE_AUX>;
		case AUDIO_FORMAT_PCM_16_BIT:
		return (AudioMixer::hook_t)
		track__NoResample<MIXTYPE_MULTI, int32_t, int16_t, int32_t>;
		return (AudioMixer::hook_t)track__NoResample<
		MIXTYPE_MULTI, int32_t /TO/, int16_t /TI/, TYPE_AUX>;
		default:
		LOG_ALWAYS_FATAL("bad mixerInFormat: %#x", mixerInFormat);
		break;
		@@ -2055,11 +2084,11 @@ AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t
		case AUDIO_FORMAT_PCM_FLOAT:
		switch (mixerOutFormat) {
		case AUDIO_FORMAT_PCM_FLOAT:
		return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
		float /TO/, float /TI/, int32_t /TA/>;
		return process_NoResampleOneTrack<
		MIXTYPE_MULTI_SAVEONLY, float /TO/, float /TI/, TYPE_AUX>;
		case AUDIO_FORMAT_PCM_16_BIT:
		return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
		int16_t, float, int32_t>;
		return process_NoResampleOneTrack<
		MIXTYPE_MULTI_SAVEONLY, int16_t /TO/, float /TI/, TYPE_AUX>;
		default:
		LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
		break;
		@@ -2068,11 +2097,11 @@ AudioMixer::process_hook_t AudioMixer::getProcessHook(int processType, uint32_t
		case AUDIO_FORMAT_PCM_16_BIT:
		switch (mixerOutFormat) {
		case AUDIO_FORMAT_PCM_FLOAT:
		return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
		float, int16_t, int32_t>;
		return process_NoResampleOneTrack<
		MIXTYPE_MULTI_SAVEONLY, float /TO/, int16_t /TI/, TYPE_AUX>;
		case AUDIO_FORMAT_PCM_16_BIT:
		return process_NoResampleOneTrack<MIXTYPE_MULTI_SAVEONLY,
		int16_t, int16_t, int32_t>;
		return process_NoResampleOneTrack<
		MIXTYPE_MULTI_SAVEONLY, int16_t /TO/, int16_t /TI/, TYPE_AUX>;
		default:
		LOG_ALWAYS_FATAL("bad mixerOutFormat: %#x", mixerOutFormat);
		break;

media/libaudioprocessing/AudioMixerOps.h

+7 −4

Original line number	Diff line number	Diff line
		@@ -188,13 +188,13 @@ inline void MixAccum(TO *auxaccum, TI value) {

		template<>
		inline void MixAccum<float, int16_t>(float *auxaccum, int16_t value) {
		static const float norm = 1. / (1 << 15);
		static constexpr float norm = 1. / (1 << 15);
		auxaccum += norm value;
		}

		template<>
		inline void MixAccum<float, int32_t>(float *auxaccum, int32_t value) {
		static const float norm = 1. / (1 << 27);
		static constexpr float norm = 1. / (1 << 27);
		auxaccum += norm value;
		}

		@@ -238,6 +238,7 @@ enum {
		* NCHAN represents number of input and output channels.
		* TO: int32_t (Q4.27) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TA: int32_t (Q4.27) or float
		* TV: int32_t (U4.28) or int16_t (U4.12) or float
		* vol: represents a volume array.
		*
		@@ -247,7 +248,8 @@ enum {
		* Single input channel. NCHAN represents number of output channels.
		* TO: int32_t (Q4.27) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TV: int32_t (U4.28) or int16_t (U4.12) or float
		* TA: int32_t (Q4.27) or float
		* TV/TAV: int32_t (U4.28) or int16_t (U4.12) or float
		* Input channel count is 1.
		* vol: represents volume array.
		*
		@@ -257,7 +259,8 @@ enum {
		* NCHAN represents number of input and output channels.
		* TO: int16_t (Q.15) or float
		* TI: int32_t (Q4.27) or int16_t (Q0.15) or float
		* TV: int32_t (U4.28) or int16_t (U4.12) or float
		* TA: int32_t (Q4.27) or float
		* TV/TAV: int32_t (U4.28) or int16_t (U4.12) or float
		* vol: represents a volume array.
		*
		* MIXTYPE_MULTI_SAVEONLY does not accumulate into the out pointer.

services/audioflinger/Configuration.h

+5 −0

Original line number	Diff line number	Diff line
		@@ -44,4 +44,9 @@
		// define FLOAT_EFFECT_CHAIN to request float effects (falls back to int16_t if unavailable)
		#define FLOAT_EFFECT_CHAIN

		#ifdef FLOAT_EFFECT_CHAIN
		// define FLOAT_AUX to process aux effect buffers in float (FLOAT_EFFECT_CHAIN must be defined)
		#define FLOAT_AUX
		#endif

		#endif // ANDROID_AUDIOFLINGER_CONFIGURATION_H

services/audioflinger/Effects.cpp

+15 −3

Original line number	Diff line number	Diff line
		@@ -342,9 +342,9 @@ void AudioFlinger::EffectModule::process()
		if (isProcessImplemented()) {
		if (auxType) {
		// We overwrite the aux input buffer here and clear after processing.
		// Note that aux input buffers are format q4_27.
		#ifdef FLOAT_EFFECT_CHAIN
		if (mSupportsFloat) {
		#ifndef FLOAT_AUX
		// Do in-place float conversion for auxiliary effect input buffer.
		static_assert(sizeof(float) <= sizeof(int32_t),
		"in-place conversion requires sizeof(float) <= sizeof(int32_t)");
		@@ -353,13 +353,21 @@ void AudioFlinger::EffectModule::process()
		mConfig.inputCfg.buffer.f32,
		mConfig.inputCfg.buffer.s32,
		mConfig.inputCfg.buffer.frameCount);
		#endif // !FLOAT_AUX
		} else
		#endif
		#endif // FLOAT_EFFECT_CHAIN
		{
		#ifdef FLOAT_AUX
		memcpy_to_i16_from_float(
		mConfig.inputCfg.buffer.s16,
		mConfig.inputCfg.buffer.f32,
		mConfig.inputCfg.buffer.frameCount);
		#else
		memcpy_to_i16_from_q4_27(
		mConfig.inputCfg.buffer.s16,
		mConfig.inputCfg.buffer.s32,
		mConfig.inputCfg.buffer.frameCount);
		#endif
		}
		}
		#ifdef FLOAT_EFFECT_CHAIN
		@@ -419,9 +427,13 @@ void AudioFlinger::EffectModule::process()

		// clear auxiliary effect input buffer for next accumulation
		if (auxType) {
		// input always q4_27 regardless of FLOAT_EFFECT_CHAIN.
		#ifdef FLOAT_AUX
		const size_t size =
		mConfig.inputCfg.buffer.frameCount * inChannelCount * sizeof(float);
		#else
		const size_t size =
		mConfig.inputCfg.buffer.frameCount * inChannelCount * sizeof(int32_t);
		#endif
		memset(mConfig.inputCfg.buffer.raw, 0, size);
		}
		} else if ((mDescriptor.flags & EFFECT_FLAG_TYPE_MASK) == EFFECT_FLAG_TYPE_INSERT &&