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Commit a97d5444 authored by Antti S. Lankila's avatar Antti S. Lankila
Browse files

Compression-related improvements 

- use a hybrid of A- and C-weighing filter to estimate loudness in compression.

- make equalizer's automatic loudness adjustment begin from 50 % midpoint
  to avoid an initial strong bass boost effect.
parent 4b07670f

media/libeffects/cyanogen-dsp/EffectCompression.cpp

+17 −15
Original line number Diff line number Diff line
@@ -51,8 +51,10 @@ int32_t EffectCompression::command(uint32_t cmdCode, uint32_t cmdSize, void* pCm 
	    return 0;

	}



	mWeighter[0].setBandPass(1700, mSamplingRate, sqrtf(2)/2);

	mWeighter[1].setBandPass(1700, mSamplingRate, sqrtf(2)/2);

        /* This filter gives a reasonable approximation of A- and C-weighting

         * which is close to correct for 100 - 10 kHz. 10 dB gain must be added to result. */

	mWeigherBP[0].setBandPass(2200, mSamplingRate, 0.33);

	mWeigherBP[1].setBandPass(2200, mSamplingRate, 0.33);



	*replyData = 0;

	return 0;
@@ -105,15 +107,16 @@ int32_t EffectCompression::command(uint32_t cmdCode, uint32_t cmdSize, void* pCm 
}



/* Return fixed point 16.48 */

uint64_t EffectCompression::estimateOneChannelLevel(audio_buffer_t *in, int32_t interleave, int32_t offset, Biquad& weighter)

uint64_t EffectCompression::estimateOneChannelLevel(audio_buffer_t *in, int32_t interleave, int32_t offset, Biquad& weigherBP)

{

    uint64_t power = 0;

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

	int32_t tmp = read(in, offset);

	offset += interleave;

        int64_t out = weighter.process(tmp);

        tmp = weigherBP.process(tmp);



	/* 2^24 * 2^24 = 48 */

        power += out * out;

        power += int64_t(tmp) * int64_t(tmp);

	offset += interleave;

    }



    return (power / in->frameCount);
@@ -124,7 +127,7 @@ int32_t EffectCompression::process_effect(audio_buffer_t *in, audio_buffer_t *ou 
    /* Analyze both channels separately, pick the maximum power measured. */

    uint64_t maximumPowerSquared = 0;

    for (uint32_t i = 0; i < mChannels; i ++) {

        uint64_t candidatePowerSquared = estimateOneChannelLevel(in, mChannels, i, mWeighter[i]);

        uint64_t candidatePowerSquared = estimateOneChannelLevel(in, mChannels, i, mWeigherBP[i]);

        if (candidatePowerSquared > maximumPowerSquared) {

            maximumPowerSquared = candidatePowerSquared;

        }
@@ -133,9 +136,8 @@ int32_t EffectCompression::process_effect(audio_buffer_t *in, audio_buffer_t *ou 
    /* -100 .. 0 dB. */

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



    /* target 83 dB SPL, and add 6 dB to compensate for the weighter, whose

     * peak is at -3 dB. */

    signalPowerDb += 96.0f - 83.0f + 6.0f;

    /* Target 83 dB SPL */

    signalPowerDb += 96.0f - 83.0f + 10.0f;



    /* now we have an estimate of the signal power, with 0 level around 83 dB.

     * we now select the level to boost to. */
@@ -159,14 +161,14 @@ int32_t EffectCompression::process_effect(audio_buffer_t *in, audio_buffer_t *ou 
        /* 8.24 */

	int32_t volAdj = desiredLevel - mCurrentLevel[i];

	

	/* I want volume adjustments to occur in about 0.05 seconds. 

	/* I want volume adjustments to occur in about 0.025 seconds. 

	 * However, if the input buffer would happen to be longer than

	 * this, I'll just make sure that I am done with the adjustment

	 * by the end of it. */

	int32_t adjLen = mSamplingRate / 20;

	/* Note: this adjustment should probably be piecewise linear

	 * approximation of an exponential to keep perceptibly linear

	 * correction rate. */

	int32_t adjLen = mSamplingRate / 40; // in practice, about 1100 frames

        /* This formulation results in piecewise linear approximation of

         * exponential because the rate of adjustment decreases from granule

         * to granule. */

	volAdj /= max(adjLen, in->frameCount);



	/* Additionally, I want volume to increase only very slowly.

media/libeffects/cyanogen-dsp/EffectCompression.h

+3 −2
Original line number Diff line number Diff line
@@ -9,9 +9,10 @@ class EffectCompression : public Effect { 
    float mCompressionRatio;

    

    int32_t mCurrentLevel[2];

    Biquad mWeighter[2];



    uint64_t estimateOneChannelLevel(audio_buffer_t *in, int32_t interleave, int32_t offset, Biquad& weighter);

    Biquad mWeigherBP[2];



    uint64_t estimateOneChannelLevel(audio_buffer_t *in, int32_t interleave, int32_t offset, Biquad& WeigherBP);



    public:

    EffectCompression();

media/libeffects/cyanogen-dsp/EffectEqualizer.cpp

+1 −1
Original line number Diff line number Diff line
@@ -61,7 +61,7 @@ static int64_t toFixedPoint(float in) { 
}



EffectEqualizer::EffectEqualizer()

    : mLoudnessAdjustment(10000.f), mLoudness(0.f)

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

{

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

        mBand[i] = 0;