equalizer: use shelfing filters for first and last band
Refactor the filter setup. Add two new filters with shelf characteristics for first and last band. Change gain calculation as recommended in the quoted document (no qrt needed). Rename variables to match the formulas in the document.
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Stefan Kost
parent
15c6175044
commit
27ea0b076a
@ -121,10 +121,6 @@ struct _GstIirEqualizerBandClass
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static GType gst_iir_equalizer_band_get_type (void);
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static void setup_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band);
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static void set_passthrough (GstIirEqualizer * equ);
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static void
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gst_iir_equalizer_band_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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@ -357,12 +353,6 @@ gst_iir_equalizer_finalize (GObject * object)
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G_OBJECT_CLASS (parent_class)->finalize (object);
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}
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static inline gdouble
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arg_to_scale (gdouble arg)
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{
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return (pow (10.0, arg / 20.0));
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}
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/* Filter taken from
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*
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* The Equivalence of Various Methods of Computing
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@ -372,34 +362,39 @@ arg_to_scale (gdouble arg)
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*
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* http://www.aes.org/e-lib/browse.cfm?elib=6326
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* http://www.musicdsp.org/files/EQ-Coefficients.pdf
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* http://www.musicdsp.org/files/Audio-EQ-Cookbook.txt
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*
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* The bandwidth method that we use here is the preferred
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* one from this article transformed from octaves to frequency
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* in Hz.
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*/
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static void
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setup_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
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static inline gdouble
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arg_to_scale (gdouble arg)
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{
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g_return_if_fail (GST_AUDIO_FILTER (equ)->format.rate);
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return (pow (10.0, arg / 40.0));
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}
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/* FIXME: we need better filters
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* - we need shelf-filter for 1st and last band
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*/
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static gdouble
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calculate_omega (gdouble freq, gint rate)
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{
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gdouble gain, omega, bw;
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gdouble edge_gain, gamma;
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gdouble alpha, beta;
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gdouble omega;
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gain = arg_to_scale (band->gain);
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if (band->freq / GST_AUDIO_FILTER (equ)->format.rate >= 0.5)
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if (freq / rate >= 0.5)
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omega = M_PI;
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else if (band->freq <= 0.0)
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else if (freq <= 0.0)
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omega = 0.0;
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else
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omega = 2.0 * M_PI * (band->freq / GST_AUDIO_FILTER (equ)->format.rate);
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omega = 2.0 * M_PI * (freq / rate);
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if (band->width / GST_AUDIO_FILTER (equ)->format.rate >= 0.5) {
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return omega;
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}
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static gdouble
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calculate_bw (GstIirEqualizerBand * band, gint rate)
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{
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gdouble bw = 0.0;
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if (band->width / rate >= 0.5) {
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/* If bandwidth == 0.5 the calculation below fails as tan(M_PI/2)
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* is undefined. So set the bandwidth to a slightly smaller value.
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*/
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@ -415,23 +410,115 @@ setup_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
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band->a2 = 0.0;
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band->b1 = 0.0;
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band->b2 = 0.0;
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gain = 1.0;
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goto out;
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} else {
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bw = 2.0 * M_PI * (band->width / GST_AUDIO_FILTER (equ)->format.rate);
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bw = 2.0 * M_PI * (band->width / rate);
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}
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return bw;
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}
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edge_gain = sqrt (gain);
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gamma = tan (bw / 2.0);
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static void
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setup_peak_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
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{
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g_return_if_fail (GST_AUDIO_FILTER (equ)->format.rate);
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alpha = gamma * edge_gain;
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beta = gamma / edge_gain;
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{
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gdouble gain, omega, bw;
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gdouble alpha, alpha1, alpha2, b0;
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gain = arg_to_scale (band->gain);
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omega = calculate_omega (band->freq, GST_AUDIO_FILTER (equ)->format.rate);
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bw = calculate_bw (band, GST_AUDIO_FILTER (equ)->format.rate);
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if (bw == 0.0)
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goto out;
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alpha = tan (bw / 2.0);
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alpha1 = alpha * gain;
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alpha2 = alpha / gain;
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b0 = (1.0 + alpha2);
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band->a0 = (1.0 + alpha1) / b0;
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band->a1 = (-2.0 * cos (omega)) / b0;
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band->a2 = (1.0 - alpha1) / b0;
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band->b1 = (2.0 * cos (omega)) / b0;
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band->b2 = -(1.0 - alpha2) / b0;
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out:
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GST_INFO
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("gain = %5.1f, width= %7.2f, freq = %7.2f, a0 = %7.5g, a1 = %7.5g, a2=%7.5g b1 = %7.5g, b2 = %7.5g",
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band->gain, band->width, band->freq, band->a0, band->a1, band->a2,
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band->b1, band->b2);
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}
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}
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static void
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setup_low_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
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{
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g_return_if_fail (GST_AUDIO_FILTER (equ)->format.rate);
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{
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gdouble gain, omega, bw;
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gdouble alpha, delta, b0;
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gdouble egp, egm;
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gain = arg_to_scale (band->gain);
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omega = calculate_omega (band->freq, GST_AUDIO_FILTER (equ)->format.rate);
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bw = calculate_bw (band, GST_AUDIO_FILTER (equ)->format.rate);
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if (bw == 0.0)
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goto out;
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egm = gain - 1.0;
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egp = gain + 1.0;
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alpha = tan (bw / 2.0);
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delta = 2.0 * sqrt (gain) * alpha;
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b0 = egp + egm * cos (omega) + delta;
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band->a0 = ((egp - egm * cos (omega) + delta) * gain) / b0;
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band->a1 = ((egm - egp * cos (omega)) * 2.0 * gain) / b0;
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band->a2 = ((egp - egm * cos (omega) - delta) * gain) / b0;
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band->b1 = ((egm + egp * cos (omega)) * 2.0) / b0;
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band->b2 = -((egp + egm * cos (omega) - delta)) / b0;
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out:
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GST_INFO
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("gain = %5.1f, width= %7.2f, freq = %7.2f, a0 = %7.5g, a1 = %7.5g, a2=%7.5g b1 = %7.5g, b2 = %7.5g",
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band->gain, band->width, band->freq, band->a0, band->a1, band->a2,
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band->b1, band->b2);
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}
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}
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static void
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setup_high_shelf_filter (GstIirEqualizer * equ, GstIirEqualizerBand * band)
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{
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g_return_if_fail (GST_AUDIO_FILTER (equ)->format.rate);
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{
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gdouble gain, omega, bw;
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gdouble alpha, delta, b0;
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gdouble egp, egm;
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gain = arg_to_scale (band->gain);
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omega = calculate_omega (band->freq, GST_AUDIO_FILTER (equ)->format.rate);
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bw = calculate_bw (band, GST_AUDIO_FILTER (equ)->format.rate);
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if (bw == 0.0)
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goto out;
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egm = gain - 1.0;
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egp = gain + 1.0;
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alpha = tan (bw / 2.0);
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delta = 2.0 * sqrt (gain) * alpha;
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b0 = egp - egm * cos (omega) + delta;
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band->a0 = ((egp + egm * cos (omega) + delta) * gain) / b0;
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band->a1 = ((egm + egp * cos (omega)) * -2.0 * gain) / b0;
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band->a2 = ((egp + egm * cos (omega) - delta) * gain) / b0;
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band->b1 = ((egm - egp * cos (omega)) * -2.0) / b0;
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band->b2 = -((egp - egm * cos (omega) - delta)) / b0;
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band->a0 = (1.0 + alpha) / (1.0 + beta);
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band->a1 = (-2.0 * cos (omega)) / (1.0 + beta);
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band->a2 = (1.0 - alpha) / (1.0 + beta);
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band->b1 = (2.0 * cos (omega)) / (1.0 + beta);
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band->b2 = -(1.0 - beta) / (1.0 + beta);
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out:
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GST_INFO
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@ -459,11 +546,14 @@ set_passthrough (GstIirEqualizer * equ)
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static void
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update_coefficients (GstIirEqualizer * equ)
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{
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gint i;
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gint i, n = equ->freq_band_count;
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for (i = 0; i < equ->freq_band_count; i++) {
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setup_filter (equ, equ->bands[i]);
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setup_low_shelf_filter (equ, equ->bands[0]);
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for (i = 1; i < n - 1; i++) {
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setup_peak_filter (equ, equ->bands[i]);
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}
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setup_high_shelf_filter (equ, equ->bands[n - 1]);
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equ->need_new_coefficients = FALSE;
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}
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