audio_utils/power.cpp - platform/system/media - Git at Google

 /*
  * Copyright 2017 The Android Open Source Project
  *
  * Licensed under the Apache License, Version 2.0 (the "License");
  * you may not use this file except in compliance with the License.
  * You may obtain a copy of the License at
  *
  *      http://www.apache.org/licenses/LICENSE-2.0
  *
  * Unless required by applicable law or agreed to in writing, software
  * distributed under the License is distributed on an "AS IS" BASIS,
  * WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
  * See the License for the specific language governing permissions and
  * limitations under the License.
  */

 // #define LOG_NDEBUG 0
 #define LOG_TAG "audio_utils_power"
 #include <log/log.h>

 #include <math.h>

 #include <audio_utils/power.h>
 #include <audio_utils/primitives.h>

 #if defined(__aarch64__) || defined(__ARM_NEON__)
 #include <arm_neon.h>
 #define USE_NEON
 #endif

 namespace {

 constexpr inline bool isFormatSupported(audio_format_t format) {
     switch (format) {
     case AUDIO_FORMAT_PCM_8_BIT:
     case AUDIO_FORMAT_PCM_16_BIT:
     case AUDIO_FORMAT_PCM_24_BIT_PACKED:
     case AUDIO_FORMAT_PCM_8_24_BIT:
     case AUDIO_FORMAT_PCM_32_BIT:
     case AUDIO_FORMAT_PCM_FLOAT:
         return true;
     default:
         return false;
     }
 }

 template <typename T>
 inline T getPtrPtrValueAndIncrement(const void **data)
 {
     return *(*reinterpret_cast<const T **>(data))++;
 }

 template <audio_format_t FORMAT>
 inline float convertToFloatAndIncrement(const void **data)
 {
     switch (FORMAT) {
     case AUDIO_FORMAT_PCM_8_BIT:
         return float_from_u8(getPtrPtrValueAndIncrement<uint8_t>(data));

     case AUDIO_FORMAT_PCM_16_BIT:
         return float_from_i16(getPtrPtrValueAndIncrement<int16_t>(data));

     case AUDIO_FORMAT_PCM_24_BIT_PACKED: {
         const uint8_t *uptr = reinterpret_cast<const uint8_t *>(*data);
         *data = uptr + 3;
         return float_from_p24(uptr);
     }

     case AUDIO_FORMAT_PCM_8_24_BIT:
         return float_from_q8_23(getPtrPtrValueAndIncrement<int32_t>(data));

     case AUDIO_FORMAT_PCM_32_BIT:
         return float_from_i32(getPtrPtrValueAndIncrement<int32_t>(data));

     case AUDIO_FORMAT_PCM_FLOAT:
         return getPtrPtrValueAndIncrement<float>(data);

     default:
         // static_assert cannot use false because the compiler may interpret it
         // even though this code path may never be taken.
         static_assert(isFormatSupported(FORMAT), "unsupported format");
     }
 }

 // used to normalize integer fixed point value to the floating point equivalent.
 template <audio_format_t FORMAT>
 constexpr inline float normalizeAmplitude()
 {
     switch (FORMAT) {
     case AUDIO_FORMAT_PCM_8_BIT:
         return 1.f / (1 << 7);

     case AUDIO_FORMAT_PCM_16_BIT:
         return 1.f / (1 << 15);

     case AUDIO_FORMAT_PCM_24_BIT_PACKED: // fall through
     case AUDIO_FORMAT_PCM_8_24_BIT:
         return 1.f / (1 << 23);

     case AUDIO_FORMAT_PCM_32_BIT:
         return 1.f / (1U << 31);

     case AUDIO_FORMAT_PCM_FLOAT:
          return 1.f;

     default:
         // static_assert cannot use false because the compiler may interpret it
         // even though this code path may never be taken.
         static_assert(isFormatSupported(FORMAT), "unsupported format");
     }
 }

 template <audio_format_t FORMAT>
 constexpr inline float normalizeEnergy()
 {
     const float val = normalizeAmplitude<FORMAT>();
     return val * val;
 }

 template <audio_format_t FORMAT>
 inline float energyMonoRef(const void *amplitudes, size_t size)
 {
     float accum(0.f);
     for (size_t i = 0; i < size; ++i) {
         const float amplitude = convertToFloatAndIncrement<FORMAT>(&amplitudes);
         accum += amplitude * amplitude;
     }
     return accum;
 }

 template <audio_format_t FORMAT>
 inline float energyMono(const void *amplitudes, size_t size)
 {
     return energyMonoRef<FORMAT>(amplitudes, size);
 }

 // fast float power computation for ARM processors that support NEON.
 #ifdef USE_NEON

 template <>
 inline float energyMono<AUDIO_FORMAT_PCM_FLOAT>(const void *amplitudes, size_t size)
 {
     float32x4_t *famplitudes = (float32x4_t *)amplitudes;

     // clear accumulator
     float32x4_t accum = vdupq_n_f32(0);

     // iterate over array getting sum of squares in 4 lanes.
     size_t i;
     for (i = 0; i < (size & ~3); i += 4) {
         accum = vmlaq_f32(accum, *famplitudes, *famplitudes);
         ++famplitudes;
     }

     // narrow 4 lanes of floats
     float32x2_t accum2 = vadd_f32(vget_low_f32(accum), vget_high_f32(accum)); // get stereo volume
     accum2 = vpadd_f32(accum2, accum2); // combine to mono

     // accumulate remainder
     float value = vget_lane_f32(accum2, 0);
     for (; i < size; ++i) {
         const float amplitude = ((float *)amplitudes)[i];
         value +=  amplitude * amplitude;
     }

     return value;
 }

 template <>
 inline float energyMono<AUDIO_FORMAT_PCM_16_BIT>(const void *amplitudes, size_t size)
 {
     int16x4_t *samplitudes = (int16x4_t *)amplitudes;

     // clear accumulator
     float32x4_t accum = vdupq_n_f32(0);

     // iterate over array getting sum of squares in 4 lanes.
     size_t i;
     for (i = 0; i < (size & ~3); i += 4) {
         // expand s16 to s32
         int32x4_t amplitude = vmovl_s16(*samplitudes);
         ++samplitudes;
         // convert s32 to f32
         float32x4_t famplitude = vcvtq_f32_s32(amplitude);
         accum = vmlaq_f32(accum, famplitude, famplitude);
     }

     // narrow 4 lanes of floats
     float32x2_t accum2 = vadd_f32(vget_low_f32(accum), vget_high_f32(accum)); // get stereo volume
     accum2 = vpadd_f32(accum2, accum2); // combine to mono

     // accumulate remainder
     float value = vget_lane_f32(accum2, 0);
     for (; i < size; ++i) {
         const float amplitude = (float)((int16_t *)amplitudes)[i];
         value +=  amplitude * amplitude;
     }

     return value * normalizeEnergy<AUDIO_FORMAT_PCM_16_BIT>();
 }

 // fast int32_t power computation for PCM_32
 template <>
 inline float energyMono<AUDIO_FORMAT_PCM_32_BIT>(const void *amplitudes, size_t size)
 {
     int32x4_t *samplitudes = (int32x4_t *)amplitudes;

     // clear accumulator
     float32x4_t accum = vdupq_n_f32(0);

     // iterate over array getting sum of squares in 4 lanes.
     size_t i;
     for (i = 0; i < (size & ~3); i += 4) {
         // convert s32 to f32
         float32x4_t famplitude = vcvtq_f32_s32(*samplitudes);
         ++samplitudes;
         accum = vmlaq_f32(accum, famplitude, famplitude);
     }

     // narrow 4 lanes of floats
     float32x2_t accum2 = vadd_f32(vget_low_f32(accum), vget_high_f32(accum)); // get stereo volume
     accum2 = vpadd_f32(accum2, accum2); // combine to mono

     // accumulate remainder
     float value = vget_lane_f32(accum2, 0);
     for (; i < size; ++i) {
         const float amplitude = (float)((int32_t *)amplitudes)[i];
         value +=  amplitude * amplitude;
     }

     return value * normalizeEnergy<AUDIO_FORMAT_PCM_32_BIT>();
 }

 // fast int32_t power computation for PCM_8_24 (essentially identical to PCM_32 above)
 template <>
 inline float energyMono<AUDIO_FORMAT_PCM_8_24_BIT>(const void *amplitudes, size_t size)
 {
     int32x4_t *samplitudes = (int32x4_t *)amplitudes;

     // clear accumulator
     float32x4_t accum = vdupq_n_f32(0);

     // iterate over array getting sum of squares in 4 lanes.
     size_t i;
     for (i = 0; i < (size & ~3); i += 4) {
         // convert s32 to f32
         float32x4_t famplitude = vcvtq_f32_s32(*samplitudes);
         ++samplitudes;
         accum = vmlaq_f32(accum, famplitude, famplitude);
     }

     // narrow 4 lanes of floats
     float32x2_t accum2 = vadd_f32(vget_low_f32(accum), vget_high_f32(accum)); // get stereo volume
     accum2 = vpadd_f32(accum2, accum2); // combine to mono

     // accumulate remainder
     float value = vget_lane_f32(accum2, 0);
     for (; i < size; ++i) {
         const float amplitude = (float)((int32_t *)amplitudes)[i];
         value +=  amplitude * amplitude;
     }

     return value * normalizeEnergy<AUDIO_FORMAT_PCM_8_24_BIT>();
 }

 #endif // USE_NEON

 } // namespace

 float audio_utils_compute_energy_mono(const void *buffer, audio_format_t format, size_t samples)
 {
     switch (format) {
     case AUDIO_FORMAT_PCM_8_BIT:
         return energyMono<AUDIO_FORMAT_PCM_8_BIT>(buffer, samples);

     case AUDIO_FORMAT_PCM_16_BIT:
         return energyMono<AUDIO_FORMAT_PCM_16_BIT>(buffer, samples);

     case AUDIO_FORMAT_PCM_24_BIT_PACKED:
         return energyMono<AUDIO_FORMAT_PCM_24_BIT_PACKED>(buffer, samples);

     case AUDIO_FORMAT_PCM_8_24_BIT:
         return energyMono<AUDIO_FORMAT_PCM_8_24_BIT>(buffer, samples);

     case AUDIO_FORMAT_PCM_32_BIT:
         return energyMono<AUDIO_FORMAT_PCM_32_BIT>(buffer, samples);

     case AUDIO_FORMAT_PCM_FLOAT:
         return energyMono<AUDIO_FORMAT_PCM_FLOAT>(buffer, samples);

     default:
         LOG_ALWAYS_FATAL("invalid format: %#x", format);
     }
 }

 float audio_utils_compute_power_mono(const void *buffer, audio_format_t format, size_t samples)
 {
     return audio_utils_power_from_energy(
             audio_utils_compute_energy_mono(buffer, format, samples) / samples);
 }

 bool audio_utils_is_compute_power_format_supported(audio_format_t format)
 {
     return isFormatSupported(format);
 }
	/*
	* Copyright 2017 The Android Open Source Project
	*
	* Licensed under the Apache License, Version 2.0 (the "License");
	* you may not use this file except in compliance with the License.
	* You may obtain a copy of the License at
	*
	* http://www.apache.org/licenses/LICENSE-2.0
	*
	* Unless required by applicable law or agreed to in writing, software
	* distributed under the License is distributed on an "AS IS" BASIS,
	* WITHOUT WARRANTIES OR CONDITIONS OF ANY KIND, either express or implied.
	* See the License for the specific language governing permissions and
	* limitations under the License.
	*/

	// #define LOG_NDEBUG 0
	#define LOG_TAG "audio_utils_power"
	#include <log/log.h>

	#include <math.h>

	#include <audio_utils/power.h>
	#include <audio_utils/primitives.h>

	#if defined(__aarch64__) \|\| defined(__ARM_NEON__)
	#include <arm_neon.h>
	#define USE_NEON
	#endif

	namespace {

	constexpr inline bool isFormatSupported(audio_format_t format) {
	switch (format) {
	case AUDIO_FORMAT_PCM_8_BIT:
	case AUDIO_FORMAT_PCM_16_BIT:
	case AUDIO_FORMAT_PCM_24_BIT_PACKED:
	case AUDIO_FORMAT_PCM_8_24_BIT:
	case AUDIO_FORMAT_PCM_32_BIT:
	case AUDIO_FORMAT_PCM_FLOAT:
	return true;
	default:
	return false;
	}
	}

	template <typename T>
	inline T getPtrPtrValueAndIncrement(const void **data)
	{
	return (reinterpret_cast<const T **>(data))++;
	}

	template <audio_format_t FORMAT>
	inline float convertToFloatAndIncrement(const void **data)
	{
	switch (FORMAT) {
	case AUDIO_FORMAT_PCM_8_BIT:
	return float_from_u8(getPtrPtrValueAndIncrement<uint8_t>(data));

	case AUDIO_FORMAT_PCM_16_BIT:
	return float_from_i16(getPtrPtrValueAndIncrement<int16_t>(data));

	case AUDIO_FORMAT_PCM_24_BIT_PACKED: {
	const uint8_t uptr = reinterpret_cast<const uint8_t >(*data);
	*data = uptr + 3;
	return float_from_p24(uptr);
	}

	case AUDIO_FORMAT_PCM_8_24_BIT:
	return float_from_q8_23(getPtrPtrValueAndIncrement<int32_t>(data));

	case AUDIO_FORMAT_PCM_32_BIT:
	return float_from_i32(getPtrPtrValueAndIncrement<int32_t>(data));

	case AUDIO_FORMAT_PCM_FLOAT:
	return getPtrPtrValueAndIncrement<float>(data);

	default:
	// static_assert cannot use false because the compiler may interpret it
	// even though this code path may never be taken.
	static_assert(isFormatSupported(FORMAT), "unsupported format");
	}
	}

	// used to normalize integer fixed point value to the floating point equivalent.
	template <audio_format_t FORMAT>
	constexpr inline float normalizeAmplitude()
	{
	switch (FORMAT) {
	case AUDIO_FORMAT_PCM_8_BIT:
	return 1.f / (1 << 7);

	case AUDIO_FORMAT_PCM_16_BIT:
	return 1.f / (1 << 15);

	case AUDIO_FORMAT_PCM_24_BIT_PACKED: // fall through
	case AUDIO_FORMAT_PCM_8_24_BIT:
	return 1.f / (1 << 23);

	case AUDIO_FORMAT_PCM_32_BIT:
	return 1.f / (1U << 31);

	case AUDIO_FORMAT_PCM_FLOAT:
	return 1.f;

	default:
	// static_assert cannot use false because the compiler may interpret it
	// even though this code path may never be taken.
	static_assert(isFormatSupported(FORMAT), "unsupported format");
	}
	}

	template <audio_format_t FORMAT>
	constexpr inline float normalizeEnergy()
	{
	const float val = normalizeAmplitude<FORMAT>();
	return val * val;
	}

	template <audio_format_t FORMAT>
	inline float energyMonoRef(const void *amplitudes, size_t size)
	{
	float accum(0.f);
	for (size_t i = 0; i < size; ++i) {
	const float amplitude = convertToFloatAndIncrement<FORMAT>(&amplitudes);
	accum += amplitude * amplitude;
	}
	return accum;
	}

	template <audio_format_t FORMAT>
	inline float energyMono(const void *amplitudes, size_t size)
	{
	return energyMonoRef<FORMAT>(amplitudes, size);
	}

	// fast float power computation for ARM processors that support NEON.
	#ifdef USE_NEON

	template <>
	inline float energyMono<AUDIO_FORMAT_PCM_FLOAT>(const void *amplitudes, size_t size)
	{
	float32x4_t famplitudes = (float32x4_t )amplitudes;

	// clear accumulator
	float32x4_t accum = vdupq_n_f32(0);

	// iterate over array getting sum of squares in 4 lanes.
	size_t i;
	for (i = 0; i < (size & ~3); i += 4) {
	accum = vmlaq_f32(accum, famplitudes, famplitudes);
	++famplitudes;
	}

	// narrow 4 lanes of floats
	float32x2_t accum2 = vadd_f32(vget_low_f32(accum), vget_high_f32(accum)); // get stereo volume
	accum2 = vpadd_f32(accum2, accum2); // combine to mono

	// accumulate remainder
	float value = vget_lane_f32(accum2, 0);
	for (; i < size; ++i) {
	const float amplitude = ((float *)amplitudes)[i];
	value += amplitude * amplitude;
	}

	return value;
	}

	template <>
	inline float energyMono<AUDIO_FORMAT_PCM_16_BIT>(const void *amplitudes, size_t size)
	{
	int16x4_t samplitudes = (int16x4_t )amplitudes;

	// clear accumulator
	float32x4_t accum = vdupq_n_f32(0);

	// iterate over array getting sum of squares in 4 lanes.
	size_t i;
	for (i = 0; i < (size & ~3); i += 4) {
	// expand s16 to s32
	int32x4_t amplitude = vmovl_s16(*samplitudes);
	++samplitudes;
	// convert s32 to f32
	float32x4_t famplitude = vcvtq_f32_s32(amplitude);
	accum = vmlaq_f32(accum, famplitude, famplitude);
	}

	// narrow 4 lanes of floats
	float32x2_t accum2 = vadd_f32(vget_low_f32(accum), vget_high_f32(accum)); // get stereo volume
	accum2 = vpadd_f32(accum2, accum2); // combine to mono

	// accumulate remainder
	float value = vget_lane_f32(accum2, 0);
	for (; i < size; ++i) {
	const float amplitude = (float)((int16_t *)amplitudes)[i];
	value += amplitude * amplitude;
	}

	return value * normalizeEnergy<AUDIO_FORMAT_PCM_16_BIT>();
	}

	// fast int32_t power computation for PCM_32
	template <>
	inline float energyMono<AUDIO_FORMAT_PCM_32_BIT>(const void *amplitudes, size_t size)
	{
	int32x4_t samplitudes = (int32x4_t )amplitudes;

	// clear accumulator
	float32x4_t accum = vdupq_n_f32(0);

	// iterate over array getting sum of squares in 4 lanes.
	size_t i;
	for (i = 0; i < (size & ~3); i += 4) {
	// convert s32 to f32
	float32x4_t famplitude = vcvtq_f32_s32(*samplitudes);
	++samplitudes;
	accum = vmlaq_f32(accum, famplitude, famplitude);
	}

	// narrow 4 lanes of floats
	float32x2_t accum2 = vadd_f32(vget_low_f32(accum), vget_high_f32(accum)); // get stereo volume
	accum2 = vpadd_f32(accum2, accum2); // combine to mono

	// accumulate remainder
	float value = vget_lane_f32(accum2, 0);
	for (; i < size; ++i) {
	const float amplitude = (float)((int32_t *)amplitudes)[i];
	value += amplitude * amplitude;
	}

	return value * normalizeEnergy<AUDIO_FORMAT_PCM_32_BIT>();
	}

	// fast int32_t power computation for PCM_8_24 (essentially identical to PCM_32 above)
	template <>
	inline float energyMono<AUDIO_FORMAT_PCM_8_24_BIT>(const void *amplitudes, size_t size)
	{
	int32x4_t samplitudes = (int32x4_t )amplitudes;

	// clear accumulator
	float32x4_t accum = vdupq_n_f32(0);

	// iterate over array getting sum of squares in 4 lanes.
	size_t i;
	for (i = 0; i < (size & ~3); i += 4) {
	// convert s32 to f32
	float32x4_t famplitude = vcvtq_f32_s32(*samplitudes);
	++samplitudes;
	accum = vmlaq_f32(accum, famplitude, famplitude);
	}

	// narrow 4 lanes of floats
	float32x2_t accum2 = vadd_f32(vget_low_f32(accum), vget_high_f32(accum)); // get stereo volume
	accum2 = vpadd_f32(accum2, accum2); // combine to mono

	// accumulate remainder
	float value = vget_lane_f32(accum2, 0);
	for (; i < size; ++i) {
	const float amplitude = (float)((int32_t *)amplitudes)[i];
	value += amplitude * amplitude;
	}

	return value * normalizeEnergy<AUDIO_FORMAT_PCM_8_24_BIT>();
	}

	#endif // USE_NEON

	} // namespace

	float audio_utils_compute_energy_mono(const void *buffer, audio_format_t format, size_t samples)
	{
	switch (format) {
	case AUDIO_FORMAT_PCM_8_BIT:
	return energyMono<AUDIO_FORMAT_PCM_8_BIT>(buffer, samples);

	case AUDIO_FORMAT_PCM_16_BIT:
	return energyMono<AUDIO_FORMAT_PCM_16_BIT>(buffer, samples);

	case AUDIO_FORMAT_PCM_24_BIT_PACKED:
	return energyMono<AUDIO_FORMAT_PCM_24_BIT_PACKED>(buffer, samples);

	case AUDIO_FORMAT_PCM_8_24_BIT:
	return energyMono<AUDIO_FORMAT_PCM_8_24_BIT>(buffer, samples);

	case AUDIO_FORMAT_PCM_32_BIT:
	return energyMono<AUDIO_FORMAT_PCM_32_BIT>(buffer, samples);

	case AUDIO_FORMAT_PCM_FLOAT:
	return energyMono<AUDIO_FORMAT_PCM_FLOAT>(buffer, samples);

	default:
	LOG_ALWAYS_FATAL("invalid format: %#x", format);
	}
	}

	float audio_utils_compute_power_mono(const void *buffer, audio_format_t format, size_t samples)
	{
	return audio_utils_power_from_energy(
	audio_utils_compute_energy_mono(buffer, format, samples) / samples);
	}

	bool audio_utils_is_compute_power_format_supported(audio_format_t format)
	{
	return isFormatSupported(format);
	}