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Commit d5a6b38b authored by Antti S. Lankila's avatar Antti S. Lankila
Browse files

Reduce loudness compensation latency and noise 

- The equalizer is now updated 100 times per second to follow the
  changes in the estimated loudness. Noise is controlled by restricting
  the adjustment to <= 0.075 dB per adjustment.

- Equalizer now has treble boost, and bass boost is slightly reduced.
parent a97d5444

media/libeffects/cyanogen-dsp/EffectEqualizer.cpp

+35 −29
Original line number Diff line number Diff line
@@ -61,7 +61,7 @@ static int64_t toFixedPoint(float in) { 
}



EffectEqualizer::EffectEqualizer()

    : mLoudnessAdjustment(10000.f), mLoudness(50.f)

    : mLoudnessAdjustment(10000.f), mLoudness(50.f), mNextUpdate(0), mNextUpdateInterval(1000), mPowerSquared(0)

{

    for (int32_t i = 0; i < 5; i ++) {

        mBand[i] = 0;
@@ -79,9 +79,10 @@ int32_t EffectEqualizer::command(uint32_t cmdCode, uint32_t cmdSize, void* pCmdD 
			return 0;

		}



                /* Weigher for estimating bass compensation. Our adjustments have range from 62.5 to 4000 Hz.

                 * Most of the adjustment is at the 62.5 Hz band, so we must concentrate on bass region. */

                mWeigher.setLowPass(1000.0, mSamplingRate, sqrtf(2)/2.);

                /* Weigher for estimating bass compensation. */

                mWeigher.setBandPass(2200.0, mSamplingRate, 0.33);

                /* 100 updates per second. */

                mNextUpdateInterval = int32_t(mSamplingRate / 100.);



		int32_t *replyData = (int32_t *) pReplyData;

		*replyData = 0;
@@ -195,10 +196,12 @@ int32_t EffectEqualizer::command(uint32_t cmdCode, uint32_t cmdSize, void* pCmdD 
 * sound pressure level.

 *

 * The boost can be calculated as linear scaling of the following adjustment:

 *   62.5 Hz +24 dB

 *    250 Hz +10 dB

 *     20 Hz +40 dB (unmodeled)

 *   62.5 Hz +20 dB

 *    250 Hz  +8 dB

 *   1000 Hz   0 dB

 *   4000 Hz -6 dB

 *   4000 Hz  -3 dB

 *  16000 Hz  +6 dB

 *

 * The boost will be applied maximally for signals of 20 dB and less,

 * and linearly decreased for signals 20 dB ... 100 dB, and no adjustment is
@@ -206,7 +209,7 @@ int32_t EffectEqualizer::command(uint32_t cmdCode, uint32_t cmdSize, void* pCmdD 
 * digital sound level against the audio.

 */

float EffectEqualizer::getAdjustedBand(int32_t band) {

    const float adj[5] = { 24.0, 10.0, 0.0, -6.0, 0.0 };

    const float adj[5] = { 20.0, 8.0, 0.0, -3.0, 6.0 };



    float loudnessLevel = mLoudness + mLoudnessAdjustment;

    if (loudnessLevel > 100.f) {
@@ -236,16 +239,35 @@ void EffectEqualizer::refreshBands() 


int32_t EffectEqualizer::process_effect(audio_buffer_t *in, audio_buffer_t *out)

{

    for (uint32_t i = 0; i < in->frameCount; i ++) {

        if (mNextUpdate == 0) {

            float signalPowerDb = logf(mPowerSquared / mNextUpdateInterval / float(int64_t(1) << 48) + 1e-10f) / logf(10.0f) * 10.0f;

            signalPowerDb += 96.0f + 10.0f;



            /* Limit automatic EQ to sub-dB adjustments to limit the noise

             * introduced by updates. */

            if (mLoudness < signalPowerDb - .5) {

                mLoudness += .5;

            } else if (mLoudness > signalPowerDb + .5) {

                mLoudness -= .5;

            } else {

                mLoudness = signalPowerDb;

            }



            /* Update EQ. */

            refreshBands();



    int64_t maximumPowerSquared = 0;

    for (uint32_t i = 0; i < in->frameCount; i ++) {

            mNextUpdate = mNextUpdateInterval;

            mPowerSquared = 0;

        }

        mNextUpdate --;



        int32_t tmpL = read(in, i * 2);

        int32_t tmpR = read(in, i * 2 + 1);



        /* Calculate signal loudness estimate */

        int64_t weight = mWeigher.process(tmpL + tmpR);

        maximumPowerSquared += weight * weight;

        mPowerSquared += weight * weight;

     

        /* first "shelve" is just gain */ 

        tmpL = tmpL * mGain >> 32;
@@ -260,22 +282,6 @@ int32_t EffectEqualizer::process_effect(audio_buffer_t *in, audio_buffer_t *out) 
        write(out, i * 2, tmpL);

        write(out, i * 2 + 1, tmpR);

    }

    maximumPowerSquared /= in->frameCount;



    float signalPowerDb = logf(maximumPowerSquared / float(int64_t(1) << 48) + 1e-10f) / logf(10.0f) * 10.0f;

    signalPowerDb += 96.0f;



    /* Limit automatic EQ to maximum of 10 dB adjustment rate per second.

     * a frame-to-frame adjusted should not be very large because adjustments do cause

     * small glitches at the output. These may be audible if single step is too large. */

    float maxAdj = in->frameCount / mSamplingRate * 10.f;

    if (mLoudness < signalPowerDb - maxAdj) {

        mLoudness += maxAdj;

    } else if (mLoudness > signalPowerDb + maxAdj) {

        mLoudness -= maxAdj;

    } else {

        mLoudness = signalPowerDb;

    }



    return 0;

}

media/libeffects/cyanogen-dsp/EffectEqualizer.h

+3 −0
Original line number Diff line number Diff line
@@ -20,6 +20,9 @@ class EffectEqualizer : public Effect { 


    Biquad mWeigher;

    float mLoudness;

    int32_t mNextUpdate;

    int32_t mNextUpdateInterval;

    int64_t mPowerSquared;



    void setBand(int32_t idx, float dB);   

    float getAdjustedBand(int32_t idx);