Mercurial > audlegacy
view Plugins/Input/musepack/equalizer.cxx @ 1695:3cf80d8a6d8d trunk
[svn] - Removed .cpp.o target since we don't have any .cpp files anymore.
- A test build then revealed that we still had two .cpp files in
Plugins/Input/musepack, fixed that.
- Moved nenolod's fix from init.mk to objective.mk.
- Removed call to ldconfig in Makefile and replaced it with a hint to
call ldconfig manually, since ldconfig without parameters breaks some
systems (mainly OpenBSD).
- I hope I did not forget to list a change I did ;)
author | js |
---|---|
date | Fri, 15 Sep 2006 06:36:26 -0700 |
parents | Plugins/Input/musepack/equalizer.cpp@f13ab2d8e9cf |
children |
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/* * Thanks to Felipe Rivera for letting us use some code from his * EQU-Plugin (http://equ.sourceforge.net/) for our Equalizer. */ #include <math.h> #include <string.h> #include <stdlib.h> #include <glib.h> #include "equalizer.h" // Fixed Point Fractional bits #define FP_FRBITS 28 // Conversions #define EQ_REAL(x) ((gint)((x) * (1 << FP_FRBITS))) /* Floating point */ typedef struct { float beta; float alpha; float gamma; } sIIRCoefficients; /* Coefficient history for the IIR filter */ typedef struct { float x[3]; /* x[n], x[n-1], x[n-2] */ float y[3]; /* y[n], y[n-1], y[n-2] */ } sXYData; /* BETA, ALPHA, GAMMA */ static sIIRCoefficients iir_cforiginal10[] = { {(9.9421504945e-01), (2.8924752745e-03), (1.9941421835e+00)}, /* 60.0 Hz */ {(9.8335039428e-01), (8.3248028618e-03), (1.9827686547e+00)}, /* 170.0 Hz */ {(9.6958094144e-01), (1.5209529281e-02), (1.9676601546e+00)}, /* 310.0 Hz */ {(9.4163923306e-01), (2.9180383468e-02), (1.9345490229e+00)}, /* 600.0 Hz */ {(9.0450844499e-01), (4.7745777504e-02), (1.8852109613e+00)}, /* 1000.0 Hz */ {(7.3940088234e-01), (1.3029955883e-01), (1.5829158753e+00)}, /* 3000.0 Hz */ {(5.4697667908e-01), (2.2651166046e-01), (1.0153238114e+00)}, /* 6000.0 Hz */ {(3.1023210589e-01), (3.4488394706e-01), (-1.8142472036e-01)}, /* 12000.0 Hz */ {(2.6718639778e-01), (3.6640680111e-01), (-5.2117742267e-01)}, /* 14000.0 Hz */ {(2.4201241845e-01), (3.7899379077e-01), (-8.0847117831e-01)}, /* 16000.0 Hz */ }; /* History for two filters */ static sXYData data_history[EQ_MAX_BANDS][EQ_CHANNELS]; static sXYData data_history2[EQ_MAX_BANDS][EQ_CHANNELS]; /* Coefficients */ static sIIRCoefficients *iir_cf; /* Gain for each band * values should be between -0.2 and 1.0 */ static float gain[10]; static float preamp; int round_trick(float floatvalue_to_round); static void output_set_eq(gboolean active, gfloat pre, gfloat * bands) { int i; preamp = 1.0 + 0.0932471 * pre + 0.00279033 * pre * pre; for (i = 0; i < 10; ++i) gain[i] = 0.03 * bands[i] + 0.000999999 * bands[i] * bands[i]; } /* Init the filter */ void init_iir(int on, float preamp_ctrl, float *eq_ctrl) { iir_cf = iir_cforiginal10; /* Zero the history arrays */ memset(data_history, 0, sizeof(sXYData) * EQ_MAX_BANDS * EQ_CHANNELS); memset(data_history2, 0, sizeof(sXYData) * EQ_MAX_BANDS * EQ_CHANNELS); output_set_eq(on, preamp_ctrl, eq_ctrl); } int iir(char *d, gint length) { gint16 *data = (gint16 *) d; /* Indexes for the history arrays * These have to be kept between calls to this function * hence they are static */ static gint i = 0, j = 2, k = 1; gint index, band, channel; gint tempgint, halflength; float out[EQ_CHANNELS], pcm[EQ_CHANNELS]; /** * IIR filter equation is * y[n] = 2 * (alpha*(x[n]-x[n-2]) + gamma*y[n-1] - beta*y[n-2]) * * NOTE: The 2 factor was introduced in the coefficients to save * a multiplication * * This algorithm cascades two filters to get nice filtering * at the expense of extra CPU cycles */ /* 16bit, 2 bytes per sample, so divide by two the length of * the buffer (length is in bytes) */ halflength = (length >> 1); for (index = 0; index < halflength; index += 2) { /* For each channel */ for (channel = 0; channel < EQ_CHANNELS; channel++) { /* No need to scale when processing the PCM with the filter */ pcm[channel] = data[index + channel]; /* Preamp gain */ pcm[channel] *= preamp; out[channel] = 0; /* For each band */ for (band = 0; band < 10; band++) { /* Store Xi(n) */ data_history[band][channel].x[i] = pcm[channel]; /* Calculate and store Yi(n) */ data_history[band][channel].y[i] = (iir_cf[band].alpha * (data_history[band][channel].x[i] - data_history[band][channel].x[k]) + iir_cf[band].gamma * data_history[band][channel].y[j] - iir_cf[band].beta * data_history[band][channel].y[k] ); /* * The multiplication by 2.0 was 'moved' into the coefficients to save * CPU cycles here */ /* Apply the gain */ out[channel] += data_history[band][channel].y[i] * gain[band]; // * 2.0; } /* For each band */ if (false) { /* Filter the sample again */ for (band = 0; band < 10; band++) { /* Store Xi(n) */ data_history2[band][channel].x[i] = out[channel]; /* Calculate and store Yi(n) */ data_history2[band][channel].y[i] = (iir_cf[band].alpha * (data_history2[band][channel].x[i] - data_history2[band][channel].x[k]) + iir_cf[band].gamma * data_history2[band][channel].y[j] - iir_cf[band].beta * data_history2[band][channel].y[k] ); /* Apply the gain */ out[channel] += data_history2[band][channel].y[i] * gain[band]; } /* For each band */ } /* Volume stuff Scale down original PCM sample and add it to the filters output. This substitutes the multiplication by 0.25 */ out[channel] += (data[index + channel] >> 2); //printf("out[channel] = %f\n", out[channel]); tempgint = (int) out[channel]; //printf("iir: old=%d new=%d\n", data[index+channel], tempgint); /* Limit the output */ if (tempgint < -32768) data[index + channel] = -32768; else if (tempgint > 32767) data[index + channel] = 32767; else data[index + channel] = tempgint; } /* For each channel */ i++; j++; k++; /* Wrap around the indexes */ if (i == 3) i = 0; else if (j == 3) j = 0; else k = 0; } /* For each pair of samples */ return length; }