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.

3 files changed, 29 insertions, 39 deletions

diff --git a/libs/audioflinger/AudioDSP.cpp b/libs/audioflinger/AudioDSP.cpp
index ae1857eb..b1c1719 100644
--- a/libs/audioflinger/AudioDSP.cpp
+++ b/libs/audioflinger/AudioDSP.cpp
@@ -329,9 +329,6 @@ 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();
 }
  
@@ -354,41 +354,30 @@ void EffectTone::setBand(int32_t band, float dB)
     refreshBands();
 }
 
-void EffectTone::refreshBands() {
+void EffectTone::refreshBands()
+{
     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];
 
-    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);
-    }
-
-    {
-        int32_t band = 3;
-
-        float dB = mBand[band + 1] - mBand[0];
-        float centerFrequency = 250.0f * powf(4, 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.
          * I'm ignoring the integer truncation problem here, but in reality
          * it should be accounted for. */
@@ -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! */
 }
diff --git a/libs/audioflinger/AudioDSP.h b/libs/audioflinger/AudioDSP.h
index 1d77a8d..e1b0933 100644
--- a/libs/audioflinger/AudioDSP.h
+++ b/libs/audioflinger/AudioDSP.h
@@ -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();
diff --git a/libs/audioflinger/AudioMixer.cpp b/libs/audioflinger/AudioMixer.cpp
index e5985d7..1853c61 100644
--- a/libs/audioflinger/AudioMixer.cpp
+++ b/libs/audioflinger/AudioMixer.cpp
@@ -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;
         }