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Commit 6ac44b2a authored by Antti S. Lankila's avatar Antti S. Lankila
Browse files

Estimate center channel and avoid DSP on it 

Also fixes reverb like I originally intended it and tweaks
parameters a little.
parent a2284ac5

libs/audioflinger/AudioDSP.cpp

+24 −6
Original line number Diff line number Diff line
@@ -406,7 +406,8 @@ void EffectHeadphone::configure(const float samplingFrequency) { 
    mReverbDelayL.setParameters(mSamplingFrequency, 0.030f);

    mReverbDelayR.setParameters(mSamplingFrequency, 0.030f);

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

    mLowpass.setHighShelf(850.0f, mSamplingFrequency, -10.0f, 0.72f);

    mLocalizationL.setHighShelf(800.0f, mSamplingFrequency, -11.0f, 0.72f);

    mLocalizationR.setHighShelf(800.0f, mSamplingFrequency, -11.0f, 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! */

}
@@ -437,6 +438,7 @@ void EffectHeadphone::process(int32_t* inout, int32_t frames) 
        /* 28 bits */

        

        if (mDeep) {

            /* Note: a pinking filter here would be good. */

            dataL += mDelayDataR;

            dataR += mDelayDataL;

        }
@@ -460,13 +462,29 @@ void EffectHeadphone::process(int32_t* inout, int32_t frames) 
        dataL += dryL;

        dataR += dryR;



        /* Lowpass filter difference to estimate head shadow. */

        int32_t diff = mLowpass.process((dataL - dataR) >> fixedPointDecimals);

        /* In matrix decoding, center channel is mixed at 0.7 and the main channel at 1.

         * It follows that the sum of them is 1.7, and the proportion of the main channel

         * must be 1 / 1.7, or about 6/10. Assuming it is so, 4/10 is the contribution

         * of center, and when 2 channels are combined, the scaler is 2/10 or 1/5.

         *

         * We could try to dynamically adjust this divisor based on cross-correlation

         * between left/right channels, which would allow us to recover a reasonable

         * estimate of the music's original center channel. */

        int32_t center = (dataL + dataR) / 5;

        int32_t directL = (dataL - center);

        int32_t directR = (dataR - center);



        /* We assume center channel reaches both ears with no coloration required.

         * We could also handle it differently at reverb stage... */



        /* Apply localization filter. */

        int32_t localizedL = mLocalizationL.process(directL >> fixedPointDecimals);

        int32_t localizedR = mLocalizationR.process(directR >> fixedPointDecimals);

        /* 28 bits */

        

        /* Mix difference between channels. */

        inout[0] = dataL - (diff >> 1);

        inout[1] = dataR + (diff >> 1);

        /* Mix difference between channels. dataX = directX + center. */

        inout[0] = dataL + localizedR;

        inout[1] = dataR + localizedL;

        inout += 2;

    }

}

libs/audioflinger/AudioDSP.h

+1 −1
Original line number Diff line number Diff line
@@ -120,7 +120,7 @@ class EffectHeadphone : public Effect { 


    Delay mReverbDelayL, mReverbDelayR;

    int32_t mDelayDataL, mDelayDataR;

    Biquad mLowpass;

    Biquad mLocalizationL, mLocalizationR;



    public:

    EffectHeadphone();