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Commit 398c4878 authored by Andy Hung's avatar Andy Hung
Browse files

LoudnessEnhancer: Add multichannel support 

Flag: EXEMPT bugfix
Test: atest loudness_enhancer_tests
Test: BT spatialization through AIDL / HIDL
Bug: 390015312
Change-Id: I152d07bb379cb42f808bc945c53682342f5175e6
parent 559e32f0

media/libeffects/loudness/EffectLoudnessEnhancer.cpp

+10 −14
Original line number Diff line number Diff line
@@ -103,7 +103,9 @@ int LE_setConfig(LoudnessEnhancerContext *pContext, effect_config_t *pConfig) 
    if (pConfig->inputCfg.samplingRate != pConfig->outputCfg.samplingRate) return -EINVAL;

    if (pConfig->inputCfg.channels != pConfig->outputCfg.channels) return -EINVAL;

    if (pConfig->inputCfg.format != pConfig->outputCfg.format) return -EINVAL;

    if (pConfig->inputCfg.channels != AUDIO_CHANNEL_OUT_STEREO) return -EINVAL;

    if (audio_channel_count_from_out_mask(pConfig->inputCfg.channels) > FCC_LIMIT) {

        return -EINVAL;

    }

    if (pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_WRITE &&

            pConfig->outputCfg.accessMode != EFFECT_BUFFER_ACCESS_ACCUMULATE) return -EINVAL;

    if (pConfig->inputCfg.format != kProcessFormat) return -EINVAL;
@@ -278,27 +280,22 @@ int LE_process( 
    }



    //ALOGV("LE about to process %d samples", inBuffer->frameCount);

    uint16_t inIdx;

    constexpr float scale = 1 << 15; // power of 2 is lossless conversion to int16_t range

    constexpr float inverseScale = 1.f / scale;

    const float inputAmp = pow(10, pContext->mTargetGainmB/2000.0f) * scale;

    float leftSample, rightSample;

    for (inIdx = 0 ; inIdx < inBuffer->frameCount ; inIdx++) {

        // makeup gain is applied on the input of the compressor

        leftSample  = inputAmp * inBuffer->f32[2*inIdx];

        rightSample = inputAmp * inBuffer->f32[2*inIdx +1];

        pContext->mCompressor->Compress(&leftSample, &rightSample);

        inBuffer->f32[2*inIdx]    = leftSample * inverseScale;

        inBuffer->f32[2*inIdx +1] = rightSample * inverseScale;

    }

    const size_t channelCount =

            audio_channel_count_from_out_mask(pContext->mConfig.outputCfg.channels);

    pContext->mCompressor->Compress(

            channelCount, inputAmp, inverseScale, inBuffer->f32, inBuffer->frameCount);



    if (inBuffer->raw != outBuffer->raw) {

        const size_t sampleCount = outBuffer->frameCount * channelCount;

        if (pContext->mConfig.outputCfg.accessMode == EFFECT_BUFFER_ACCESS_ACCUMULATE) {

            for (size_t i = 0; i < outBuffer->frameCount*2; i++) {

            for (size_t i = 0; i < sampleCount; i++) {

                outBuffer->f32[i] += inBuffer->f32[i];

            }

        } else {

            memcpy(outBuffer->raw, inBuffer->raw, outBuffer->frameCount * 2 * sizeof(float));

            memcpy(outBuffer->raw, inBuffer->raw, sampleCount * sizeof(float));

        }

    }

    if (pContext->mState != LOUDNESS_ENHANCER_STATE_ACTIVE) {
@@ -462,4 +459,3 @@ audio_effect_library_t AUDIO_EFFECT_LIBRARY_INFO_SYM = { 
};



}; // extern "C"

media/libeffects/loudness/aidl/EffectLoudnessEnhancer.cpp

+2 −1
Original line number Diff line number Diff line
@@ -139,7 +139,8 @@ std::shared_ptr<EffectContext> LoudnessEnhancerImpl::createContext( 
        LOG(DEBUG) << __func__ << " context already exist";

        return mContext;

    }

    if (2 < getChannelCount(common.input.base.channelMask)) {

    const int channelCount = getChannelCount(common.input.base.channelMask);

    if (FCC_LIMIT < channelCount) {

        LOG(ERROR) << __func__

                   << " channelCount not supported: " << common.input.base.channelMask.toString();

        return nullptr;

media/libeffects/loudness/aidl/LoudnessEnhancerContext.cpp

+6 −25
Original line number Diff line number Diff line
@@ -69,34 +69,15 @@ IEffect::Status LoudnessEnhancerContext::process(float* in, float* out, int samp 
    constexpr float scale = 1 << 15;  // power of 2 is lossless conversion to int16_t range

    constexpr float inverseScale = 1.f / scale;

    const float inputAmp = pow(10, mGain / 2000.0f) * scale;

    float leftSample, rightSample;



    if (mCompressor != nullptr) {

        for (int inIdx = 0; inIdx < samples; inIdx += 2) {

            // makeup gain is applied on the input of the compressor

            leftSample = inputAmp * in[inIdx];

            rightSample = inputAmp * in[inIdx + 1];

            mCompressor->Compress(&leftSample, &rightSample);

            in[inIdx] = leftSample * inverseScale;

            in[inIdx + 1] = rightSample * inverseScale;

        }

    } else {

        for (int inIdx = 0; inIdx < samples; inIdx += 2) {

            leftSample = inputAmp * in[inIdx];

            rightSample = inputAmp * in[inIdx + 1];

            in[inIdx] = leftSample * inverseScale;

            in[inIdx + 1] = rightSample * inverseScale;

        }

        const size_t channelCount = aidl::android::hardware::audio::common::getChannelCount(

                mCommon.input.base.channelMask);

        const size_t frameCount = samples / channelCount;

        mCompressor->Compress(channelCount, inputAmp, inverseScale, in, frameCount);

    }

    bool accumulate = false;

    if (in != out) {

        for (int i = 0; i < samples; i++) {

            if (accumulate) {

                out[i] += in[i];

            } else {

                out[i] = in[i];

            }

        }

        // nit: update Compress() to write to out.

        memcpy(out, in, samples * sizeof(float));

    }

    return {STATUS_OK, samples, samples};

}

media/libeffects/loudness/dsp/core/dynamic_range_compression.cpp

+45 −57
Original line number Diff line number Diff line
@@ -22,6 +22,7 @@ 
#include "dsp/core/basic.h"

#include "dsp/core/interpolation.h"

#include "dsp/core/dynamic_range_compression.h"

#include <system/audio.h>



#include <android/log.h>
@@ -74,64 +75,51 @@ bool AdaptiveDynamicRangeCompression::Initialize( 
  return true;

}



float AdaptiveDynamicRangeCompression::Compress(float x) {

  const float max_abs_x = std::max(std::fabs(x), kMinLogAbsValue);

  const float max_abs_x_dB = math::fast_log(max_abs_x);

  // Subtract Threshold from log-encoded input to get the amount of overshoot

  const float overshoot = max_abs_x_dB - knee_threshold_;

  // Hard half-wave rectifier

  const float rect = std::max(overshoot, 0.0f);

  // Multiply rectified overshoot with slope

  const float cv = rect * slope_;

  const float prev_state = state_;

  if (cv <= state_) {

    state_ = alpha_attack_ * state_ + (1.0f - alpha_attack_) * cv;

  } else {

    state_ = alpha_release_ * state_ + (1.0f - alpha_release_) * cv;

  }

  compressor_gain_ *= expf(state_ - prev_state);

  x *= compressor_gain_;

  if (x > kFixedPointLimit) {

    return kFixedPointLimit;

  }

  if (x < -kFixedPointLimit) {

    return -kFixedPointLimit;

  }

  return x;

}



void AdaptiveDynamicRangeCompression::Compress(float *x1, float *x2) {

  // Taking the maximum amplitude of both channels

  const float max_abs_x = std::max(std::fabs(*x1),

    std::max(std::fabs(*x2), kMinLogAbsValue));

  const float max_abs_x_dB = math::fast_log(max_abs_x);

  // Subtract Threshold from log-encoded input to get the amount of overshoot

  const float overshoot = max_abs_x_dB - knee_threshold_;

  // Hard half-wave rectifier

  const float rect = std::max(overshoot, 0.0f);

  // Multiply rectified overshoot with slope

  const float cv = rect * slope_;

  const float prev_state = state_;

  if (cv <= state_) {

    state_ = alpha_attack_ * state_ + (1.0f - alpha_attack_) * cv;

  } else {

    state_ = alpha_release_ * state_ + (1.0f - alpha_release_) * cv;

  }

  compressor_gain_ *= expf(state_ - prev_state);

  *x1 *= compressor_gain_;

  if (*x1 > kFixedPointLimit) {

    *x1 = kFixedPointLimit;

  }

  if (*x1 < -kFixedPointLimit) {

    *x1 = -kFixedPointLimit;

  }

  *x2 *= compressor_gain_;

  if (*x2 > kFixedPointLimit) {

    *x2 = kFixedPointLimit;

  }

  if (*x2 < -kFixedPointLimit) {

    *x2 = -kFixedPointLimit;

// Instantiate Compress for supported channel counts.

#define INSTANTIATE_COMPRESS(CHANNEL_COUNT) \

case CHANNEL_COUNT: \

    if constexpr (CHANNEL_COUNT <= FCC_LIMIT) { \

        Compress(inputAmp, inverseScale, \

                reinterpret_cast<internal_array_t<float, CHANNEL_COUNT>*>(buffer), frameCount); \

        return; \

    } \

    break;



 void AdaptiveDynamicRangeCompression::Compress(size_t channelCount,

        float inputAmp, float inverseScale, float* buffer, size_t frameCount) {

    using android::audio_utils::intrinsics::internal_array_t;

    switch (channelCount) {

        INSTANTIATE_COMPRESS(1);

        INSTANTIATE_COMPRESS(2);

        INSTANTIATE_COMPRESS(3);

        INSTANTIATE_COMPRESS(4);

        INSTANTIATE_COMPRESS(5);

        INSTANTIATE_COMPRESS(6);

        INSTANTIATE_COMPRESS(7);

        INSTANTIATE_COMPRESS(8);

        INSTANTIATE_COMPRESS(9);

        INSTANTIATE_COMPRESS(10);

        INSTANTIATE_COMPRESS(11);

        INSTANTIATE_COMPRESS(12);

        INSTANTIATE_COMPRESS(13);

        INSTANTIATE_COMPRESS(14);

        INSTANTIATE_COMPRESS(15);

        INSTANTIATE_COMPRESS(16);

        INSTANTIATE_COMPRESS(17);

        INSTANTIATE_COMPRESS(18);

        INSTANTIATE_COMPRESS(19);

        INSTANTIATE_COMPRESS(20);

        INSTANTIATE_COMPRESS(21);

        INSTANTIATE_COMPRESS(22);

        INSTANTIATE_COMPRESS(23);

        INSTANTIATE_COMPRESS(24);

        INSTANTIATE_COMPRESS(25);

        INSTANTIATE_COMPRESS(26);

        INSTANTIATE_COMPRESS(27);

        INSTANTIATE_COMPRESS(28);

    }

    LOG_ALWAYS_FATAL("%s: channelCount: %zu not supported", __func__, channelCount);

}



}  // namespace le_fx

media/libeffects/loudness/dsp/core/dynamic_range_compression.h

+32 −13
Original line number Diff line number Diff line
@@ -23,6 +23,7 @@ 
#include "dsp/core/basic.h"

#include "dsp/core/interpolation.h"



#include <audio_utils/intrinsic_utils.h>

#include <android/log.h>



namespace le_fx {
@@ -51,19 +52,9 @@ class AdaptiveDynamicRangeCompression { 
    // relatively safe choice for many signals.

    bool Initialize(float target_gain, float sampling_rate);



  // A fast version of the algorithm that uses approximate computations for the

  // log(.) and exp(.).

  float Compress(float x);



  // Stereo channel version of the compressor

  void Compress(float *x1, float *x2);



  // This version is slower than Compress(.) but faster than CompressSlow(.)

  float CompressNormalSpeed(float x);



  // A slow version of the algorithm that is easier for further developement,

  // tuning and debugging

  float CompressSlow(float x);

  // in-place compression.

  void Compress(size_t channelCount,

          float inputAmp, float inverseScale, float* buffer, size_t frameCount);



  // Sets knee threshold (in decibel).

  void set_knee_threshold(float decibel);
@@ -73,6 +64,34 @@ class AdaptiveDynamicRangeCompression { 
  void set_knee_threshold_via_target_gain(float target_gain);



 private:



  // Templated Compress routine.

  template <typename V>

  void Compress(float inputAmp, float inverseScale, V* buffer, size_t frameCount) {

    for (size_t i = 0; i < frameCount; ++i) {

        auto v = android::audio_utils::intrinsics::vmul(buffer[i], inputAmp);

        const float max_abs_x = android::audio_utils::intrinsics::vmaxv(

                android::audio_utils::intrinsics::vabs(v));

        const float max_abs_x_dB = math::fast_log(std::max(max_abs_x, kMinLogAbsValue));

        // Subtract Threshold from log-encoded input to get the amount of overshoot

        const float overshoot = max_abs_x_dB - knee_threshold_;

        // Hard half-wave rectifier

        const float rect = std::max(overshoot, 0.0f);

        // Multiply rectified overshoot with slope

        const float cv = rect * slope_;

        const float prev_state = state_;

        if (cv <= state_) {

            state_ = alpha_attack_ * state_ + (1.0f - alpha_attack_) * cv;

        } else {

            state_ = alpha_release_ * state_ + (1.0f - alpha_release_) * cv;

        }

        compressor_gain_ *= expf(state_ - prev_state);

        const auto x = android::audio_utils::intrinsics::vmul(v, compressor_gain_);

        v = android::audio_utils::intrinsics::vclamp(x, -kFixedPointLimit, kFixedPointLimit);

        buffer[i] = android::audio_utils::intrinsics::vmul(inverseScale, v);

    }

  }



  // The minimum accepted absolute input value and it's natural logarithm. This

  // is to prevent numerical issues when the input is close to zero

  static const float kMinAbsValue;