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Commit e49f5811 authored by Antti S. Lankila's avatar Antti S. Lankila
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Various DSP enhancements 

- Update headset to a 2nd order shelf filter, drop allpass delays

  A single shelf filter can give a time delay for low frequencies
  and attenuation for high frequencies yielding a headphone virtualization
  directly.

- Make equalizer out of high shelves.

  Cascading 4 high shelves and one global gain gives me 5 controllable bands
  with passthrough characteristics when not used.

- Set loudness target to 93 %

Also correcting a volume-ramping bug which sometimes caused a slow
adjustment to desired volume because of integer truncation and invalid
round down/up algorithm. This should eliminate cases where song starts
super loud.
parent 2246292e

libs/audioflinger/AudioDSP.cpp

+21 −33
Original line number Diff line number Diff line
@@ -328,9 +328,6 @@ float EffectCompression::estimateLevel(const int16_t* audioData, int32_t frames, 


EffectTone::EffectTone()

{

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

        mBand[i] = 0;

    }

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

        setBand(i, 0);

    }
@@ -345,6 +342,9 @@ EffectTone::~EffectTone() { 


void EffectTone::configure(const float samplingFrequency) {

    Effect::configure(samplingFrequency);

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

        mBand[i] = 0;

    }

    refreshBands();

}
@@ -354,39 +354,28 @@ void EffectTone::setBand(int32_t band, float dB) 
    refreshBands();

}



void EffectTone::refreshBands() {

    mGain = toFixedPoint(powf(10, mBand[0] / 20));



    for (int32_t band = 0; band < 3; band ++) {

        float dB = mBand[band + 1] - mBand[0];

        float centerFrequency = 250.0f * powf(4, band);



        mFilterL[band].setPeakingEqualizer(centerFrequency, mSamplingFrequency, dB, 3.0f);

        mFilterR[band].setPeakingEqualizer(centerFrequency, mSamplingFrequency, dB, 3.0f);

    }



void EffectTone::refreshBands()

{

        int32_t band = 3;



        float dB = mBand[band + 1] - mBand[0];

        float centerFrequency = 250.0f * powf(4, band);

    mGain = toFixedPoint(powf(10, mBand[0] / 20));

    for (int band = 0; band < 4; band ++) {

        float centerFrequency = 62.5f * powf(4, band);

        float dB = mBand[band+1] - mBand[band];



        mFilterL[band].setHighShelf(centerFrequency * 0.5f, mSamplingFrequency, dB, 1.0f);

        mFilterR[band].setHighShelf(centerFrequency * 0.5f, mSamplingFrequency, dB, 1.0f);

        mFilterL[band].setHighShelf(centerFrequency * 2.0f, mSamplingFrequency, dB, 1.0f);

        mFilterR[band].setHighShelf(centerFrequency * 2.0f, mSamplingFrequency, dB, 1.0f);

    }

}



void EffectTone::process(int32_t* inout, int32_t frames)

{

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

        int32_t tmpL = inout[0] >> fixedPointDecimals;

        int32_t tmpR = inout[1] >> fixedPointDecimals;

        /* 16 bits */

       

        /* bass control is really a global gain compensated by other

         * controls */

        tmpL = tmpL * mGain;

        tmpR = tmpR * mGain;

        int32_t tmpL = inout[0];

        int32_t tmpR = inout[1];

        /* 28 bits */

     

        /* first "shelve" is just gain */ 

        tmpL = (tmpL >> fixedPointDecimals) * mGain;

        tmpR = (tmpR >> fixedPointDecimals) * mGain;

        /* 28 bits */

 

        /* evaluate the other filters.
@@ -424,10 +413,9 @@ void EffectHeadphone::configure(const float samplingFrequency) { 


    mReverbDelayL.setParameters(mSamplingFrequency, 0.030f);

    mReverbDelayR.setParameters(mSamplingFrequency, 0.030f);

    /* the -3 dB point is around 700 Hz, similar to the classic design

     * in bs2b. */

    mLowpassL.setHighShelf1(1500.0f, mSamplingFrequency, -16.0f);

    mLowpassR.setHighShelf1(1500.0f, mSamplingFrequency, -16.0f);

    /* the -3 dB point is around 650 Hz, giving about 300 us to work with */

    mLowpassL.setHighShelf(800.0f, mSamplingFrequency, -12.0f, 0.72f);

    mLowpassR.setHighShelf(800.0f, mSamplingFrequency, -12.0f, 0.72f);

    /* Rockbox has a 0.3 ms delay line (13 samples at 44100 Hz), but

     * I think it makes the whole effect sound pretty bad so I skipped it! */

}

libs/audioflinger/AudioDSP.h

+1 −1
Original line number Diff line number Diff line
@@ -101,7 +101,7 @@ class EffectCompression : public Effect { 


class EffectTone : public Effect {

    float mBand[5];

    int32_t mGain;

    int mGain;

    Biquad mFilterL[4], mFilterR[4];



    void refreshBands();

libs/audioflinger/AudioMixer.cpp

+7 −5
Original line number Diff line number Diff line
@@ -38,8 +38,8 @@ static inline int16_t clamp16(int32_t sample) 
    return sample;

}



static int32_t seed = 1;

inline static int32_t prng() {

    static int32_t seed = 1;

    seed = (seed * 12345) + 1103515245;

    return int32_t(seed & 0xfff);

}
@@ -59,6 +59,8 @@ AudioMixer::AudioMixer(size_t frameCount, uint32_t sampleRate, AudioDSP& dsp) 
    track_t* t = mState.tracks;

    for (int i=0 ; i<32 ; i++) {

        t->needs = 0;

        t->prevVolume[0] = UNITY_GAIN;

        t->prevVolume[1] = UNITY_GAIN;

        t->volume[0] = UNITY_GAIN;

        t->volume[1] = UNITY_GAIN;

        t->volumeInc[0] = 0;
@@ -255,17 +257,17 @@ void AudioMixer::track_t::adjustVolumeRamp(AudioDSP& dsp, size_t frames) 
        int32_t d = desiredVolume - prevVolume[i];



        /* limit change rate to smooth the compressor. */

        int32_t volChangeLimit = (prevVolume[i] >> 10);

        int32_t volChangeLimit = (prevVolume[i] >> 9);



        volChangeLimit += 1;

        volChangeLimit -= 1;

        int32_t volInc = d / int32_t(frames);

        if (volInc < -(volChangeLimit)) {

            volInc = -(volChangeLimit);

        }



        /* Make ramps up slower, but ramps down fast. */

        volChangeLimit >>= 3;

        volChangeLimit -= 1;

        volChangeLimit >>= 4;

        volChangeLimit += 1;

        if (volInc > volChangeLimit) {

            volInc = volChangeLimit;

        }