//taken from Vortex
void AnalyzeSound()
{
int m_fps = 60;
// sum (left channel) spectrum up into 3 bands
// [note: the new ranges do it so that the 3 bands are equally spaced, pitch-wise]
float min_freq = 200.0f;
float max_freq = 11025.0f;
float net_octaves = (logf(max_freq / min_freq) / logf(2.0f)); // 5.7846348455575205777914165223593
float octaves_per_band = net_octaves / 3.0f; // 1.9282116151858401925971388407864
float mult = powf(2.0f, octaves_per_band); // each band's highest freq. divided by its lowest freq.; 3.805831305510122517035102576162
// [to verify: min_freq * mult * mult * mult should equal max_freq.]
// for (int ch=0; ch<2; ch++)
{
for (int i = 0; i<3; i++)
{
// old guesswork code for this:
// float exp = 2.1f;
// int start = (int)(NUM_FREQUENCIES*0.5f*powf(i/3.0f, exp));
// int end = (int)(NUM_FREQUENCIES*0.5f*powf((i+1)/3.0f, exp));
// results:
// old range: new range (ideal):
// bass: 0-1097 200-761
// mids: 1097-4705 761-2897
// treb: 4705-11025 2897-11025
int start = (int)(NUM_FREQUENCIES * min_freq*powf(mult, (float)i) / 11025.0f);
int end = (int)(NUM_FREQUENCIES * min_freq*powf(mult, (float)i + 1) / 11025.0f);
if (start < 0) start = 0;
if (end > NUM_FREQUENCIES) end = NUM_FREQUENCIES;
g_sound.imm[0][i] = 0;
for (int j = start; j<end; j++)
{
g_sound.imm[0][i] += g_sound.fSpectrum[0][j];
g_sound.imm[0][i] += g_sound.fSpectrum[1][j];
}
g_sound.imm[0][i] /= (float)(end - start) * 2;
}
}
// multiply by long-term, empirically-determined inverse averages:
// (for a trial of 244 songs, 10 seconds each, somewhere in the 2nd or 3rd minute,
// the average levels were: 0.326781557 0.38087377 0.199888934
for (int ch = 0; ch<2; ch++)
{
g_sound.imm[ch][0] /= 0.326781557f;//0.270f;
g_sound.imm[ch][1] /= 0.380873770f;//0.343f;
g_sound.imm[ch][2] /= 0.199888934f;//0.295f;
}
// do temporal blending to create attenuated and super-attenuated versions
for (int ch = 0; ch<2; ch++)
{
for (int i = 0; i<3; i++)
{
// g_sound.avg[i]
{
float avg_mix;
if (g_sound.imm[ch][i] > g_sound.avg[ch][i])
avg_mix = AdjustRateToFPS(0.2f, 14.0f, (float)m_fps);
else
avg_mix = AdjustRateToFPS(0.5f, 14.0f, (float)m_fps);
// if (g_sound.imm[ch][i] > g_sound.avg[ch][i])
// avg_mix = 0.5f;
// else
// avg_mix = 0.8f;
g_sound.avg[ch][i] = g_sound.avg[ch][i] * avg_mix + g_sound.imm[ch][i] * (1 - avg_mix);
}
{
float med_mix = 0.91f;//0.800f + 0.11f*powf(t, 0.4f); // primarily used for velocity_damping
float long_mix = 0.96f;//0.800f + 0.16f*powf(t, 0.2f); // primarily used for smoke plumes
med_mix = AdjustRateToFPS(med_mix, 14.0f, (float)m_fps);
long_mix = AdjustRateToFPS(long_mix, 14.0f, (float)m_fps);
g_sound.med_avg[ch][i] = g_sound.med_avg[ch][i] * (med_mix)+g_sound.imm[ch][i] * (1 - med_mix);
// g_sound.long_avg[ch][i] = g_sound.long_avg[ch][i]*(long_mix) + g_sound.imm[ch][i]*(1-long_mix);
}
}
}
float newBass = ((g_sound.avg[0][0] - g_sound.med_avg[0][0]) / g_sound.med_avg[0][0]) * 2;
float newMiddle = ((g_sound.avg[0][1] - g_sound.med_avg[0][1]) / g_sound.med_avg[0][1]) * 2;
float newTreble = ((g_sound.avg[0][2] - g_sound.med_avg[0][2]) / g_sound.med_avg[0][2]) * 2;
newBass = std::max(std::min(newBass, 1.0f), -1.0f);
newMiddle = std::max(std::min(newMiddle, 1.0f), -1.0f);
newTreble = std::max(std::min(newTreble, 1.0f), -1.0f);
g_bassLast = g_bass;
g_middleLast = g_middle;
float avg_mix;
if (newTreble > g_treble)
avg_mix = 0.5f;
else
avg_mix = 0.5f;
//dealing with NaN's in linux
if (g_bass != g_bass) g_bass = 0;
if (g_middle != g_middle) g_middle = 0;
if (g_treble != g_treble) g_treble = 0;
g_bass = g_bass*avg_mix + newBass*(1 - avg_mix);
g_middle = g_middle*avg_mix + newMiddle*(1 - avg_mix);
//g_treble = g_treble*avg_mix + newTreble*(1 - avg_mix);
g_bass = std::max(std::min(g_bass, 1.0f), -1.0f);
g_middle = std::max(std::min(g_middle, 1.0f), -1.0f);
//g_treble = std::max(std::min(g_treble, 1.0f), -1.0f);
if (g_middle < 0) g_middle = g_middle * -1.0f;
if (g_bass < 0) g_bass = g_bass * -1.0f;
if (((g_middle - g_middleLast) > beatThreshold ||
(g_bass - g_bassLast > beatThreshold))
&& ((fAppTime - fLightTime) > 0.3f))
{
//beat
FastBeatLights();
CycleHue(1500);
//changed lights
fLightTime = fAppTime;
}
}
void AudioVis::AnalyzeNewSound(long rsize)
{
// we get 576 samples in from winamp.
// the output of the fft has 'num_frequencies' samples,
// and represents the frequency range 0 hz - 22,050 hz.
// usually, plugins only use half of this output (the range 0 hz - 11,025 hz),
// since >10 khz doesn't usually contribute much.
int i;
float temp_wave[2][576];
int samp=0;
int old_i = 0;
AudioVisData *data = this->data;
AudioVis *rd = this;
FFT *m_fftobj = data->fft;
for (i=0; i<576; i++, samp+=2)
{
// rd->fWaveform[0][i] = (float)(((float)C816(rd->cWaveformL[i]))/32768.0f);
// rd->fWaveform[1][i] = (float)(((float)C816(rd->cWaveformR[i]))/32768.0f);
rd->fWaveform[0][i] = (float)((rd->cWaveformL[i]) -128);
rd->fWaveform[1][i] = (float)((rd->cWaveformR[i]) -128);
// fWaveform[0][i] = (float)((pWave[samp]/(0x8000)));
// fWaveform[1][i] = (float)((pWave[samp+1]/(0x8000)));
// simulating single frequencies from 200 to 11,025 Hz:
//float freq = 1.0f + 11050*(GetFrame() % 100)*0.01f;
//rd->fWaveform[0][i] = 10*sinf(i*freq*6.28f/44100.0f);
// damp the input into the FFT a bit, to reduce high-frequency noise:
temp_wave[0][i] = 0.5f*(rd->fWaveform[0][i] + rd->fWaveform[0][old_i]);
temp_wave[1][i] = 0.5f*(rd->fWaveform[1][i] + rd->fWaveform[1][old_i]);
old_i = i;
}
m_fftobj->time_to_frequency_domain(temp_wave[0], rd->fSpectrum[0]);
m_fftobj->time_to_frequency_domain(temp_wave[1], rd->fSpectrum[1]);
// sum (left channel) spectrum up into 3 bands
// [note: the new ranges do it so that the 3 bands are equally spaced, pitch-wise]
float min_freq = 60.0f;
float max_freq = 11025.0f;
float net_octaves = (logf(max_freq/min_freq) / logf(2.0f)); // 5.7846348455575205777914165223593
float octaves_per_band = net_octaves / 16.0f; // 1.9282116151858401925971388407864
float mult = powf(2.0f, octaves_per_band); // each band's highest freq. divided by its lowest freq.; 3.805831305510122517035102576162
// [to verify: min_freq * mult * mult * mult should equal max_freq.]
for (int ch=0; ch<2; ch++)
{
for (i=0; i<16; i++)
{
// old guesswork code for this:
// float exp = 2.1f;
// int start = (int)(NUM_FREQUENCIES*0.5f*powf(i/3.0f, exp));
// int end = (int)(NUM_FREQUENCIES*0.5f*powf((i+1)/3.0f, exp));
// results:
// old range: new range (ideal):
// bass: 0-1097 200-761
// mids: 1097-4705 761-2897
// treb: 4705-11025 2897-11025
int start = (int)(NUM_FREQUENCIES * min_freq*powf(mult, i )/11025.0f);
int end = (int)(NUM_FREQUENCIES * min_freq*powf(mult, i+1)/11025.0f);
if (start < 0) start = 0;
if (end > NUM_FREQUENCIES) end = NUM_FREQUENCIES;
rd->imm[ch][i] = 0;
for (int j=start; j<end; j++)
rd->imm[ch][i] += rd->fSpectrum[ch][j];
rd->imm[ch][i] /= (float)(end-start);
}
}
// do temporal blending to create attenuated and super-attenuated versions
for (int ch=0; ch<2; ch++)
{
for (i=0; i<16; i++)
{
// rd->avg[i]
{
float avg_mix;
if (rd->imm[ch][i] > rd->avg[ch][i])
avg_mix = AdjustRateToFPS(0.2f, 76.0f, m_fps);
else
avg_mix = AdjustRateToFPS(0.5f, 76.0f, m_fps);
rd->avg[ch][i] = rd->avg[ch][i]*avg_mix + rd->imm[ch][i]*(1-avg_mix);
}
// rd->med_avg[i]
// rd->long_avg[i]
{
float med_mix = 0.91f;//0.800f + 0.11f*powf(t, 0.4f); // primarily used for velocity_damping
float long_mix = 0.96f;//0.800f + 0.16f*powf(t, 0.2f); // primarily used for smoke plumes
med_mix = AdjustRateToFPS( med_mix, 76.0f, m_fps);
long_mix = AdjustRateToFPS(long_mix, 76.0f, m_fps);
rd->med_avg[ch][i] = rd->med_avg[ch][i]*(med_mix ) + rd->imm[ch][i]*(1-med_mix );
rd->long_avg[ch][i] = rd->long_avg[ch][i]*(long_mix) + rd->imm[ch][i]*(1-long_mix);
}
}
}
BeatDetect();
this->updateCount++;
}
