From 842451a72045b962c008c93f32f52a53aba1eb42 Mon Sep 17 00:00:00 2001 From: Sebastian Dröge Date: Thu, 16 Aug 2007 17:02:07 +0000 Subject: gst/audiofx/: Add Chebyshev lowpass/highpass and bandpass/bandreject elements. Original commit message from CVS: reviewed by: Stefan Kost * gst/audiofx/Makefile.am: * gst/audiofx/audiochebyshevfreqband.c: (gst_audio_chebyshev_freq_band_mode_get_type), (gst_audio_chebyshev_freq_band_base_init), (gst_audio_chebyshev_freq_band_dispose), (gst_audio_chebyshev_freq_band_class_init), (gst_audio_chebyshev_freq_band_init), (generate_biquad_coefficients), (calculate_gain), (generate_coefficients), (gst_audio_chebyshev_freq_band_set_property), (gst_audio_chebyshev_freq_band_get_property), (gst_audio_chebyshev_freq_band_setup), (process), (process_64), (process_32), (gst_audio_chebyshev_freq_band_transform_ip), (gst_audio_chebyshev_freq_band_start): * gst/audiofx/audiochebyshevfreqband.h: * gst/audiofx/audiochebyshevfreqlimit.c: (gst_audio_chebyshev_freq_limit_mode_get_type), (gst_audio_chebyshev_freq_limit_base_init), (gst_audio_chebyshev_freq_limit_dispose), (gst_audio_chebyshev_freq_limit_class_init), (gst_audio_chebyshev_freq_limit_init), (generate_biquad_coefficients), (calculate_gain), (generate_coefficients), (gst_audio_chebyshev_freq_limit_set_property), (gst_audio_chebyshev_freq_limit_get_property), (gst_audio_chebyshev_freq_limit_setup), (process), (process_64), (process_32), (gst_audio_chebyshev_freq_limit_transform_ip), (gst_audio_chebyshev_freq_limit_start): * gst/audiofx/audiochebyshevfreqlimit.h: * gst/audiofx/audiofx.c: (plugin_init): Add Chebyshev lowpass/highpass and bandpass/bandreject elements. Fixes #464800. * tests/check/Makefile.am: * tests/check/elements/.cvsignore: * tests/check/elements/audiochebyshevfreqband.c: (setup_audiochebyshevfreqband), (cleanup_audiochebyshevfreqband), (GST_START_TEST), (audiochebyshevfreqband_suite), (main): * tests/check/elements/audiochebyshevfreqlimit.c: (setup_audiochebyshevfreqlimit), (cleanup_audiochebyshevfreqlimit), (GST_START_TEST), (audiochebyshevfreqlimit_suite), (main): Add unit tests for the chebyshev filters. * docs/plugins/Makefile.am: * docs/plugins/gst-plugins-good-plugins-docs.sgml: * docs/plugins/gst-plugins-good-plugins-sections.txt: * docs/plugins/gst-plugins-good-plugins.args: * docs/plugins/inspect/plugin-1394.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-dv.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-jpeg.xml: * docs/plugins/inspect/plugin-png.xml: * docs/plugins/inspect/plugin-rtp.xml: * docs/plugins/inspect/plugin-shout2send.xml: * docs/plugins/inspect/plugin-wavpack.xml: And add docs for the chebyshev filters. While doing that also run make update in docs/plugins. --- gst/audiofx/Makefile.am | 11 +- gst/audiofx/audiochebband.c | 916 ++++++++++++++++++++++++++++++++++ gst/audiofx/audiochebband.h | 79 +++ gst/audiofx/audiocheblimit.c | 816 ++++++++++++++++++++++++++++++ gst/audiofx/audiocheblimit.h | 78 +++ gst/audiofx/audiochebyshevfreqband.c | 916 ++++++++++++++++++++++++++++++++++ gst/audiofx/audiochebyshevfreqband.h | 79 +++ gst/audiofx/audiochebyshevfreqlimit.c | 816 ++++++++++++++++++++++++++++++ gst/audiofx/audiochebyshevfreqlimit.h | 78 +++ gst/audiofx/audiofx.c | 8 +- 10 files changed, 3793 insertions(+), 4 deletions(-) create mode 100644 gst/audiofx/audiochebband.c create mode 100644 gst/audiofx/audiochebband.h create mode 100644 gst/audiofx/audiocheblimit.c create mode 100644 gst/audiofx/audiocheblimit.h create mode 100644 gst/audiofx/audiochebyshevfreqband.c create mode 100644 gst/audiofx/audiochebyshevfreqband.h create mode 100644 gst/audiofx/audiochebyshevfreqlimit.c create mode 100644 gst/audiofx/audiochebyshevfreqlimit.h (limited to 'gst/audiofx') diff --git a/gst/audiofx/Makefile.am b/gst/audiofx/Makefile.am index b5bf0bf9..61a8dcc8 100644 --- a/gst/audiofx/Makefile.am +++ b/gst/audiofx/Makefile.am @@ -7,7 +7,9 @@ libgstaudiofx_la_SOURCES = audiofx.c\ audiopanorama.c \ audioinvert.c \ audioamplify.c \ - audiodynamic.c + audiodynamic.c \ + audiochebyshevfreqlimit.c \ + audiochebyshevfreqband.c # flags used to compile this plugin libgstaudiofx_la_CFLAGS = $(GST_CFLAGS) \ @@ -18,12 +20,15 @@ libgstaudiofx_la_LIBADD = $(GST_LIBS) \ $(GST_BASE_LIBS) \ $(GST_CONTROLLER_LIBS) \ $(GST_PLUGINS_BASE_LIBS) \ - -lgstaudio-$(GST_MAJORMINOR) + -lgstaudio-$(GST_MAJORMINOR) \ + $(LIBM) libgstaudiofx_la_LDFLAGS = $(GST_PLUGIN_LDFLAGS) # headers we need but don't want installed noinst_HEADERS = audiopanorama.h \ audioinvert.h \ audioamplify.h \ - audiodynamic.h + audiodynamic.h \ + audiochebyshevfreqlimit.h \ + audiochebyshevfreqband.c diff --git a/gst/audiofx/audiochebband.c b/gst/audiofx/audiochebband.c new file mode 100644 index 00000000..d4730607 --- /dev/null +++ b/gst/audiofx/audiochebband.c @@ -0,0 +1,916 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/* + * Chebyshev type 1 filter design based on + * "The Scientist and Engineer's Guide to DSP", Chapter 20. + * http://www.dspguide.com/ + * + * For type 2 and Chebyshev filters in general read + * http://en.wikipedia.org/wiki/Chebyshev_filter + * + * Transformation from lowpass to bandpass/bandreject: + * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm + * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm + * + */ + +/** + * SECTION:element-audiochebyshevfreqband + * @short_description: Chebyshev band pass and band reject filter + * + * + * + * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency + * band. The number of poles and the ripple parameter control the rolloff. + * + * + * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. + * some frequencies in the passband will be amplified by that value. A higher ripple value will allow + * a faster rolloff. + * + * + * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will + * be at most this value. A lower ripple value will allow a faster rolloff. + * + * + * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. + * + * Example launch line + * + * + * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink + * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include + +#include "audiochebyshevfreqband.h" + +#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand", + "Filter/Effect/Audio", + "Chebyshev band pass and band reject filter", + "Sebastian Dröge "); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_MODE, + PROP_TYPE, + PROP_LOWER_FREQUENCY, + PROP_UPPER_FREQUENCY, + PROP_RIPPLE, + PROP_POLES +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width = (int) { 32, 64 }, " \ + " endianness = (int) BYTE_ORDER," \ + " rate = (int) [ 1, MAX ]," \ + " channels = (int) [ 1, MAX ]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element"); + +GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band, + GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); + +static void gst_audio_chebyshev_freq_band_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_audio_chebyshev_freq_band_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter, + GstRingBufferSpec * format); +static GstFlowReturn +gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base); + +static void process_64 (GstAudioChebyshevFreqBand * filter, + gdouble * data, guint num_samples); +static void process_32 (GstAudioChebyshevFreqBand * filter, + gfloat * data, guint num_samples); + +enum +{ + MODE_BAND_PASS = 0, + MODE_BAND_REJECT +}; + +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ()) +static GType +gst_audio_chebyshev_freq_band_mode_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {MODE_BAND_PASS, "Band pass (default)", + "band-pass"}, + {MODE_BAND_REJECT, "Band reject", + "band-reject"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values); + } + return gtype; +} + +/* GObject vmethod implementations */ + +static void +gst_audio_chebyshev_freq_band_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details (element_class, &element_details); + + caps = gst_caps_from_string (ALLOWED_CAPS); + gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), + caps); + gst_caps_unref (caps); +} + +static void +gst_audio_chebyshev_freq_band_dispose (GObject * object) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass * + klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + GstAudioFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property; + gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property; + gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose; + + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", + "Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE, + MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_TYPE, + g_param_spec_int ("type", "Type", + "Type of the chebychev filter", 1, 2, + 1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY, + g_param_spec_float ("lower-frequency", "Lower frequency", + "Start frequency of the band (Hz)", 0.0, G_MAXFLOAT, + 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY, + g_param_spec_float ("upper-frequency", "Upper frequency", + "Stop frequency of the band (Hz)", 0.0, G_MAXFLOAT, + 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_RIPPLE, + g_param_spec_float ("ripple", "Ripple", + "Amount of ripple (dB)", 0.0, G_MAXFLOAT, + 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_POLES, + g_param_spec_int ("poles", "Poles", + "Number of poles to use, will be rounded up to the next multiply of four", + 4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup); + trans_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start); +} + +static void +gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter, + GstAudioChebyshevFreqBandClass * klass) +{ + filter->lower_frequency = filter->upper_frequency = 0.0; + filter->mode = MODE_BAND_PASS; + filter->type = 1; + filter->poles = 4; + filter->ripple = 0.25; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); + + filter->have_coeffs = FALSE; + filter->num_a = 0; + filter->num_b = 0; + filter->channels = NULL; +} + +static void +generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter, + gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3, + gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4) +{ + gint np = filter->poles / 2; + gdouble ripple = filter->ripple; + + /* pole location in s-plane */ + gdouble rp, ip; + + /* zero location in s-plane */ + gdouble rz = 0.0, iz = 0.0; + + /* transfer function coefficients for the z-plane */ + gdouble x0, x1, x2, y1, y2; + gint type = filter->type; + + /* Calculate pole location for lowpass at frequency 1 */ + { + gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + + rp = -sin (angle); + ip = cos (angle); + } + + /* If we allow ripple, move the pole from the unit + * circle to an ellipse and keep cutoff at frequency 1 */ + if (ripple > 0 && type == 1) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + + vx = (1.0 / np) * asinh (1.0 / es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } else if (type == 2) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + vx = (1.0 / np) * asinh (es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } + + /* Calculate inverse of the pole location to move from + * type I to type II */ + if (type == 2) { + gdouble mag2 = rp * rp + ip * ip; + + rp /= mag2; + ip /= mag2; + } + + /* Calculate zero location for frequency 1 on the + * unit circle for type 2 */ + if (type == 2) { + gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble mag2; + + rz = 0.0; + iz = cos (angle); + mag2 = rz * rz + iz * iz; + rz /= mag2; + iz /= mag2; + } + + /* Convert from s-domain to z-domain by + * using the bilinear Z-transform, i.e. + * substitute s by (2/t)*((z-1)/(z+1)) + * with t = 2 * tan(0.5). + */ + if (type == 1) { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t) / d; + x1 = 2.0 * x0; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } else { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t * iz * iz + 4.0) / d; + x1 = (-8.0 + 2.0 * iz * iz * t * t) / d; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } + + /* Convert from lowpass at frequency 1 to either bandpass + * or band reject. + * + * For bandpass substitute z^(-1) with: + * + * -2 -1 + * -z + alpha * z - beta + * ---------------------------- + * -2 -1 + * beta * z - alpha * z + 1 + * + * alpha = (2*a*b)/(1+b) + * beta = (b-1)/(b+1) + * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) + * b = tan(1/2) * cot((w1 - w0)/2) + * + * For bandreject substitute z^(-1) with: + * + * -2 -1 + * z - alpha * z + beta + * ---------------------------- + * -2 -1 + * beta * z - alpha * z + 1 + * + * alpha = (2*a)/(1+b) + * beta = (1-b)/(1+b) + * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) + * b = tan(1/2) * tan((w1 - w0)/2) + * + */ + { + gdouble a, b, d; + gdouble alpha, beta; + gdouble w0 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w1 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + + if (filter->mode == MODE_BAND_PASS) { + a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); + b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0); + + alpha = (2.0 * a * b) / (1.0 + b); + beta = (b - 1.0) / (b + 1.0); + + d = 1.0 + beta * (y1 - beta * y2); + + *a0 = (x0 + beta * (-x1 + beta * x2)) / d; + *a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d; + *a2 = + (-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) + + alpha * alpha * (x0 - x1 + x2)) / d; + *a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d; + *a4 = (beta * (beta * x0 - x1) + x2) / d; + *b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d; + *b2 = + (-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) + + 2.0 * beta * (-1.0 + y2)) / d; + *b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d; + *b4 = (-beta * beta - beta * y1 + y2) / d; + } else { + a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); + b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0); + + alpha = (2.0 * a) / (1.0 + b); + beta = (1.0 - b) / (1.0 + b); + + d = -1.0 + beta * (beta * y2 + y1); + + *a0 = (-x0 - beta * x1 - beta * beta * x2) / d; + *a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d; + *a2 = + (-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) - + alpha * alpha * (x0 + x1 + x2)) / d; + *a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d; + *a4 = (-beta * beta * x0 - beta * x1 - x2) / d; + *b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d; + *b2 = + -(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) + + alpha * alpha * (-1.0 + y1 + y2)) / d; + *b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d; + *b4 = -(-beta * beta + beta * y1 + y2) / d; + } + } +} + +/* Evaluate the transfer function that corresponds to the IIR + * coefficients at zr + zi*I and return the magnitude */ +static gdouble +calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr, + gdouble zi) +{ + gdouble sum_ar, sum_ai; + gdouble sum_br, sum_bi; + gdouble gain_r, gain_i; + + gdouble sum_r_old; + gdouble sum_i_old; + + gint i; + + sum_ar = 0.0; + sum_ai = 0.0; + for (i = num_a; i >= 0; i--) { + sum_r_old = sum_ar; + sum_i_old = sum_ai; + + sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; + sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; + } + + sum_br = 0.0; + sum_bi = 0.0; + for (i = num_b; i >= 0; i--) { + sum_r_old = sum_br; + sum_i_old = sum_bi; + + sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; + sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; + } + sum_br += 1.0; + sum_bi += 0.0; + + gain_r = + (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + gain_i = + (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + + return (sqrt (gain_r * gain_r + gain_i * gain_i)); +} + +static void +generate_coefficients (GstAudioChebyshevFreqBand * filter) +{ + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + if (GST_AUDIO_FILTER (filter)->format.rate == 0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + GST_LOG_OBJECT (filter, "rate was not set yet"); + return; + } + + filter->have_coeffs = TRUE; + + if (filter->upper_frequency <= filter->lower_frequency) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + GST_LOG_OBJECT (filter, "frequency band had no or negative dimension"); + return; + } + + if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) { + filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2; + GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency"); + } + + if (filter->lower_frequency < 0.0) { + filter->lower_frequency = 0.0; + GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0"); + } + + /* Calculate coefficients for the chebyshev filter */ + { + gint np = filter->poles; + gdouble *a, *b; + gint i, p; + + filter->num_a = np + 1; + filter->a = a = g_new0 (gdouble, np + 5); + filter->num_b = np + 1; + filter->b = b = g_new0 (gdouble, np + 5); + + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + for (i = 0; i < channels; i++) { + GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i]; + + ctx->x = g_new0 (gdouble, np + 1); + ctx->y = g_new0 (gdouble, np + 1); + } + + /* Calculate transfer function coefficients */ + a[4] = 1.0; + b[4] = 1.0; + + for (p = 1; p <= np / 4; p++) { + gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4; + gdouble *ta = g_new0 (gdouble, np + 5); + gdouble *tb = g_new0 (gdouble, np + 5); + + generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1, + &b2, &b3, &b4); + + memcpy (ta, a, sizeof (gdouble) * (np + 5)); + memcpy (tb, b, sizeof (gdouble) * (np + 5)); + + /* add the new coefficients for the new two poles + * to the cascade by multiplication of the transfer + * functions */ + for (i = 4; i < np + 5; i++) { + a[i] = + a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] + + a4 * ta[i - 4]; + b[i] = + tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] - + b4 * tb[i - 4]; + } + g_free (ta); + g_free (tb); + } + + /* Move coefficients to the beginning of the array + * and multiply the b coefficients with -1 to move from + * the transfer function's coefficients to the difference + * equation's coefficients */ + b[4] = 0.0; + for (i = 0; i <= np; i++) { + a[i] = a[i + 4]; + b[i] = -b[i + 4]; + } + + /* Normalize to unity gain at frequency 0 and frequency + * 0.5 for bandreject and unity gain at band center frequency + * for bandpass */ + if (filter->mode == MODE_BAND_REJECT) { + /* gain is sqrt(H(0)*H(0.5)) */ + + gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0); + gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0); + + gain1 = sqrt (gain1 * gain2); + + for (i = 0; i <= np; i++) { + a[i] /= gain1; + } + } else { + /* gain is H(wc), wc = center frequency */ + + gdouble w1 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w2 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w0 = (w2 + w1) / 2.0; + gdouble zr = cos (w0), zi = sin (w0); + gdouble gain = calculate_gain (a, b, np, np, zr, zi); + + for (i = 0; i <= np; i++) { + a[i] /= gain; + } + } + + GST_LOG_OBJECT (filter, + "Generated IIR coefficients for the Chebyshev filter"); + GST_LOG_OBJECT (filter, + "mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB", + (filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject", + filter->type, filter->poles, filter->lower_frequency, + filter->upper_frequency, filter->ripple); + + GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0))); + { + gdouble w1 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w2 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w0 = (w2 + w1) / 2.0; + gdouble zr, zi; + + zr = cos (w1); + zi = sin (w1); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->lower_frequency); + zr = cos (w0); + zi = sin (w0); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) ((filter->lower_frequency + filter->upper_frequency) / 2.0)); + zr = cos (w2); + zi = sin (w2); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->upper_frequency); + } + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)), + GST_AUDIO_FILTER (filter)->format.rate / 2); + } +} + +static void +gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + switch (prop_id) { + case PROP_MODE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->mode = g_value_get_enum (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_TYPE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->type = g_value_get_int (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_LOWER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (filter); + filter->lower_frequency = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_UPPER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (filter); + filter->upper_frequency = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_RIPPLE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->ripple = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_POLES: + GST_BASE_TRANSFORM_LOCK (filter); + filter->poles = GST_ROUND_UP_4 (g_value_get_int (value)); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + switch (prop_id) { + case PROP_MODE: + g_value_set_enum (value, filter->mode); + break; + case PROP_TYPE: + g_value_set_int (value, filter->type); + break; + case PROP_LOWER_FREQUENCY: + g_value_set_float (value, filter->lower_frequency); + break; + case PROP_UPPER_FREQUENCY: + g_value_set_float (value, filter->upper_frequency); + break; + case PROP_RIPPLE: + g_value_set_float (value, filter->ripple); + break; + case PROP_POLES: + g_value_set_int (value, filter->poles); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + gboolean ret = TRUE; + + if (format->width == 32) + filter->process = (GstAudioChebyshevFreqBandProcessFunc) + process_32; + else if (format->width == 64) + filter->process = (GstAudioChebyshevFreqBandProcessFunc) + process_64; + else + ret = FALSE; + + filter->have_coeffs = FALSE; + + return ret; +} + +static inline gdouble +process (GstAudioChebyshevFreqBand * filter, + GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0) +{ + gdouble val = filter->a[0] * x0; + gint i, j; + + for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) { + val += filter->a[i] * ctx->x[j]; + j--; + if (j < 0) + j = filter->num_a - 1; + } + + for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) { + val += filter->b[i] * ctx->y[j]; + j--; + if (j < 0) + j = filter->num_b - 1; + } + + if (ctx->x) { + ctx->x_pos++; + if (ctx->x_pos > filter->num_a - 1) + ctx->x_pos = 0; + ctx->x[ctx->x_pos] = x0; + } + + if (ctx->y) { + ctx->y_pos++; + if (ctx->y_pos > filter->num_b - 1) + ctx->y_pos = 0; + + ctx->y[ctx->y_pos] = val; + } + + return val; +} + +static void +process_64 (GstAudioChebyshevFreqBand * filter, + gdouble * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +static void +process_32 (GstAudioChebyshevFreqBand * filter, + gfloat * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, + GstBuffer * buf) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); + + if (!filter->have_coeffs) + generate_coefficients (filter); + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} + +static gboolean +gst_audio_chebyshev_freq_band_start (GstBaseTransform * base) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i; + + /* Reset the history of input and output values if + * already existing */ + if (channels && filter->channels) { + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + if (ctx->x) + memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble)); + if (ctx->y) + memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble)); + } + } + return TRUE; +} diff --git a/gst/audiofx/audiochebband.h b/gst/audiofx/audiochebband.h new file mode 100644 index 00000000..e8c58074 --- /dev/null +++ b/gst/audiofx/audiochebband.h @@ -0,0 +1,79 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ +#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND (gst_audio_chebyshev_freq_band_get_type()) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) +typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand; +typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass; + +typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint); + +typedef struct +{ + gdouble *x; + gint x_pos; + gdouble *y; + gint y_pos; +} GstAudioChebyshevFreqBandChannelCtx; + +struct _GstAudioChebyshevFreqBand +{ + GstAudioFilter audiofilter; + + gint mode; + gint type; + gint poles; + gfloat lower_frequency; + gfloat upper_frequency; + gfloat ripple; + + /* < private > */ + GstAudioChebyshevFreqBandProcessFunc process; + + gboolean have_coeffs; + gdouble *a; + gint num_a; + gdouble *b; + gint num_b; + GstAudioChebyshevFreqBandChannelCtx *channels; +}; + +struct _GstAudioChebyshevFreqBandClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_chebyshev_freq_band_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */ diff --git a/gst/audiofx/audiocheblimit.c b/gst/audiofx/audiocheblimit.c new file mode 100644 index 00000000..872b277d --- /dev/null +++ b/gst/audiofx/audiocheblimit.c @@ -0,0 +1,816 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/* + * Chebyshev type 1 filter design based on + * "The Scientist and Engineer's Guide to DSP", Chapter 20. + * http://www.dspguide.com/ + * + * For type 2 and Chebyshev filters in general read + * http://en.wikipedia.org/wiki/Chebyshev_filter + * + */ + +/** + * SECTION:element-audiochebyshevfreqlimit + * @short_description: Chebyshev low pass and high pass filter + * + * + * + * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the + * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff. + * + * + * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. + * some frequencies in the passband will be amplified by that value. A higher ripple value will allow + * a faster rolloff. + * + * + * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will + * be at most this value. A lower ripple value will allow a faster rolloff. + * + * + * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. + * + * Example launch line + * + * + * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink + * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include + +#include "audiochebyshevfreqlimit.h" + +#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit", + "Filter/Effect/Audio", + "Chebyshev low pass and high pass filter", + "Sebastian Dröge "); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_MODE, + PROP_TYPE, + PROP_CUTOFF, + PROP_RIPPLE, + PROP_POLES +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width = (int) { 32, 64 }, " \ + " endianness = (int) BYTE_ORDER," \ + " rate = (int) [ 1, MAX ]," \ + " channels = (int) [ 1, MAX ]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element"); + +GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit, + gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, + DEBUG_INIT); + +static void gst_audio_chebyshev_freq_limit_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_audio_chebyshev_freq_limit_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter, + GstRingBufferSpec * format); +static GstFlowReturn +gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base); + +static void process_64 (GstAudioChebyshevFreqLimit * filter, + gdouble * data, guint num_samples); +static void process_32 (GstAudioChebyshevFreqLimit * filter, + gfloat * data, guint num_samples); + +enum +{ + MODE_LOW_PASS = 0, + MODE_HIGH_PASS +}; + +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ()) +static GType +gst_audio_chebyshev_freq_limit_mode_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {MODE_LOW_PASS, "Low pass (default)", + "low-pass"}, + {MODE_HIGH_PASS, "High pass", + "high-pass"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values); + } + return gtype; +} + +/* GObject vmethod implementations */ + +static void +gst_audio_chebyshev_freq_limit_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details (element_class, &element_details); + + caps = gst_caps_from_string (ALLOWED_CAPS); + gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), + caps); + gst_caps_unref (caps); +} + +static void +gst_audio_chebyshev_freq_limit_dispose (GObject * object) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass * + klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + GstAudioFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property; + gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property; + gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose; + + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", + "Low pass or high pass mode", + GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_TYPE, + g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_CUTOFF, + g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0, + G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_RIPPLE, + g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0, + G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_POLES, + g_param_spec_int ("poles", "Poles", + "Number of poles to use, will be rounded up to the next even number", + 2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + filter_class->setup = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup); + trans_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start); +} + +static void +gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter, + GstAudioChebyshevFreqLimitClass * klass) +{ + filter->cutoff = 0.0; + filter->mode = MODE_LOW_PASS; + filter->type = 1; + filter->poles = 4; + filter->ripple = 0.25; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); + + filter->have_coeffs = FALSE; + filter->num_a = 0; + filter->num_b = 0; + filter->channels = NULL; +} + +static void +generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter, + gint p, gdouble * a0, gdouble * a1, gdouble * a2, + gdouble * b1, gdouble * b2) +{ + gint np = filter->poles; + gdouble ripple = filter->ripple; + + /* pole location in s-plane */ + gdouble rp, ip; + + /* zero location in s-plane */ + gdouble rz = 0.0, iz = 0.0; + + /* transfer function coefficients for the z-plane */ + gdouble x0, x1, x2, y1, y2; + gint type = filter->type; + + /* Calculate pole location for lowpass at frequency 1 */ + { + gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + + rp = -sin (angle); + ip = cos (angle); + } + + /* If we allow ripple, move the pole from the unit + * circle to an ellipse and keep cutoff at frequency 1 */ + if (ripple > 0 && type == 1) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + + vx = (1.0 / np) * asinh (1.0 / es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } else if (type == 2) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + vx = (1.0 / np) * asinh (es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } + + /* Calculate inverse of the pole location to convert from + * type I to type II */ + if (type == 2) { + gdouble mag2 = rp * rp + ip * ip; + + rp /= mag2; + ip /= mag2; + } + + /* Calculate zero location for frequency 1 on the + * unit circle for type 2 */ + if (type == 2) { + gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble mag2; + + rz = 0.0; + iz = cos (angle); + mag2 = rz * rz + iz * iz; + rz /= mag2; + iz /= mag2; + } + + /* Convert from s-domain to z-domain by + * using the bilinear Z-transform, i.e. + * substitute s by (2/t)*((z-1)/(z+1)) + * with t = 2 * tan(0.5). + */ + if (type == 1) { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t) / d; + x1 = 2.0 * x0; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } else { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t * iz * iz + 4.0) / d; + x1 = (-8.0 + 2.0 * iz * iz * t * t) / d; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } + + /* Convert from lowpass at frequency 1 to either lowpass + * or highpass. + * + * For lowpass substitute z^(-1) with: + * -1 + * z - k + * ------------ + * -1 + * 1 - k * z + * + * k = sin((1-w)/2) / sin((1+w)/2) + * + * For highpass substitute z^(-1) with: + * + * -1 + * -z - k + * ------------ + * -1 + * 1 + k * z + * + * k = -cos((1+w)/2) / cos((1-w)/2) + * + */ + { + gdouble k, d; + gdouble omega = + 2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate); + + if (filter->mode == MODE_LOW_PASS) + k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0); + else + k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0); + + d = 1.0 + y1 * k - y2 * k * k; + *a0 = (x0 + k * (-x1 + k * x2)) / d; + *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d; + *a2 = (x0 * k * k - x1 * k + x2) / d; + *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d; + *b2 = (-k * k - y1 * k + y2) / d; + + if (filter->mode == MODE_HIGH_PASS) { + *a1 = -*a1; + *b1 = -*b1; + } + } +} + +/* Evaluate the transfer function that corresponds to the IIR + * coefficients at zr + zi*I and return the magnitude */ +static gdouble +calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr, + gdouble zi) +{ + gdouble sum_ar, sum_ai; + gdouble sum_br, sum_bi; + gdouble gain_r, gain_i; + + gdouble sum_r_old; + gdouble sum_i_old; + + gint i; + + sum_ar = 0.0; + sum_ai = 0.0; + for (i = num_a; i >= 0; i--) { + sum_r_old = sum_ar; + sum_i_old = sum_ai; + + sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; + sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; + } + + sum_br = 0.0; + sum_bi = 0.0; + for (i = num_b; i >= 0; i--) { + sum_r_old = sum_br; + sum_i_old = sum_bi; + + sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; + sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; + } + sum_br += 1.0; + sum_bi += 0.0; + + gain_r = + (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + gain_i = + (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + + return (sqrt (gain_r * gain_r + gain_i * gain_i)); +} + +static void +generate_coefficients (GstAudioChebyshevFreqLimit * filter) +{ + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + if (GST_AUDIO_FILTER (filter)->format.rate == 0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "rate was not set yet"); + return; + } + + filter->have_coeffs = TRUE; + + if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency"); + return; + } else if (filter->cutoff <= 0.0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "cutoff is lower than zero"); + return; + } + + /* Calculate coefficients for the chebyshev filter */ + { + gint np = filter->poles; + gdouble *a, *b; + gint i, p; + + filter->num_a = np + 1; + filter->a = a = g_new0 (gdouble, np + 3); + filter->num_b = np + 1; + filter->b = b = g_new0 (gdouble, np + 3); + + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + for (i = 0; i < channels; i++) { + GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i]; + + ctx->x = g_new0 (gdouble, np + 1); + ctx->y = g_new0 (gdouble, np + 1); + } + + /* Calculate transfer function coefficients */ + a[2] = 1.0; + b[2] = 1.0; + + for (p = 1; p <= np / 2; p++) { + gdouble a0, a1, a2, b1, b2; + gdouble *ta = g_new0 (gdouble, np + 3); + gdouble *tb = g_new0 (gdouble, np + 3); + + generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2); + + memcpy (ta, a, sizeof (gdouble) * (np + 3)); + memcpy (tb, b, sizeof (gdouble) * (np + 3)); + + /* add the new coefficients for the new two poles + * to the cascade by multiplication of the transfer + * functions */ + for (i = 2; i < np + 3; i++) { + a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2]; + b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2]; + } + g_free (ta); + g_free (tb); + } + + /* Move coefficients to the beginning of the array + * and multiply the b coefficients with -1 to move from + * the transfer function's coefficients to the difference + * equation's coefficients */ + b[2] = 0.0; + for (i = 0; i <= np; i++) { + a[i] = a[i + 2]; + b[i] = -b[i + 2]; + } + + /* Normalize to unity gain at frequency 0 for lowpass + * and frequency 0.5 for highpass */ + { + gdouble gain; + + if (filter->mode == MODE_LOW_PASS) + gain = calculate_gain (a, b, np, np, 1.0, 0.0); + else + gain = calculate_gain (a, b, np, np, -1.0, 0.0); + + for (i = 0; i <= np; i++) { + a[i] /= gain; + } + } + + GST_LOG_OBJECT (filter, + "Generated IIR coefficients for the Chebyshev filter"); + GST_LOG_OBJECT (filter, + "mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB", + (filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass", + filter->type, filter->poles, filter->cutoff, filter->ripple); + GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0))); + { + gdouble wc = + 2.0 * M_PI * (filter->cutoff / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble zr = cos (wc), zi = sin (wc); + + GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->cutoff); + } + GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)), + GST_AUDIO_FILTER (filter)->format.rate / 2); + } +} + +static void +gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + switch (prop_id) { + case PROP_MODE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->mode = g_value_get_enum (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_TYPE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->type = g_value_get_int (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_CUTOFF: + GST_BASE_TRANSFORM_LOCK (filter); + filter->cutoff = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_RIPPLE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->ripple = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_POLES: + GST_BASE_TRANSFORM_LOCK (filter); + filter->poles = GST_ROUND_UP_2 (g_value_get_int (value)); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + switch (prop_id) { + case PROP_MODE: + g_value_set_enum (value, filter->mode); + break; + case PROP_TYPE: + g_value_set_int (value, filter->type); + break; + case PROP_CUTOFF: + g_value_set_float (value, filter->cutoff); + break; + case PROP_RIPPLE: + g_value_set_float (value, filter->ripple); + break; + case PROP_POLES: + g_value_set_int (value, filter->poles); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + gboolean ret = TRUE; + + if (format->width == 32) + filter->process = (GstAudioChebyshevFreqLimitProcessFunc) + process_32; + else if (format->width == 64) + filter->process = (GstAudioChebyshevFreqLimitProcessFunc) + process_64; + else + ret = FALSE; + + filter->have_coeffs = FALSE; + + return ret; +} + +static inline gdouble +process (GstAudioChebyshevFreqLimit * filter, + GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0) +{ + gdouble val = filter->a[0] * x0; + gint i, j; + + for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) { + val += filter->a[i] * ctx->x[j]; + j--; + if (j < 0) + j = filter->num_a - 1; + } + + for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) { + val += filter->b[i] * ctx->y[j]; + j--; + if (j < 0) + j = filter->num_b - 1; + } + + if (ctx->x) { + ctx->x_pos++; + if (ctx->x_pos > filter->num_a - 1) + ctx->x_pos = 0; + ctx->x[ctx->x_pos] = x0; + } + + if (ctx->y) { + ctx->y_pos++; + if (ctx->y_pos > filter->num_b - 1) + ctx->y_pos = 0; + + ctx->y[ctx->y_pos] = val; + } + + return val; +} + +static void +process_64 (GstAudioChebyshevFreqLimit * filter, + gdouble * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +static void +process_32 (GstAudioChebyshevFreqLimit * filter, + gfloat * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, + GstBuffer * buf) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); + + if (!filter->have_coeffs) + generate_coefficients (filter); + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} + + +static gboolean +gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i; + + /* Reset the history of input and output values if + * already existing */ + if (channels && filter->channels) { + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + if (ctx->x) + memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble)); + if (ctx->y) + memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble)); + } + } + return TRUE; +} diff --git a/gst/audiofx/audiocheblimit.h b/gst/audiofx/audiocheblimit.h new file mode 100644 index 00000000..4c87ba8e --- /dev/null +++ b/gst/audiofx/audiocheblimit.h @@ -0,0 +1,78 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ +#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT (gst_audio_chebyshev_freq_limit_get_type()) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) +typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit; +typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass; + +typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint); + +typedef struct +{ + gdouble *x; + gint x_pos; + gdouble *y; + gint y_pos; +} GstAudioChebyshevFreqLimitChannelCtx; + +struct _GstAudioChebyshevFreqLimit +{ + GstAudioFilter audiofilter; + + gint mode; + gint type; + gint poles; + gfloat cutoff; + gfloat ripple; + + /* < private > */ + GstAudioChebyshevFreqLimitProcessFunc process; + + gboolean have_coeffs; + gdouble *a; + gint num_a; + gdouble *b; + gint num_b; + GstAudioChebyshevFreqLimitChannelCtx *channels; +}; + +struct _GstAudioChebyshevFreqLimitClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_chebyshev_freq_limit_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */ diff --git a/gst/audiofx/audiochebyshevfreqband.c b/gst/audiofx/audiochebyshevfreqband.c new file mode 100644 index 00000000..d4730607 --- /dev/null +++ b/gst/audiofx/audiochebyshevfreqband.c @@ -0,0 +1,916 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/* + * Chebyshev type 1 filter design based on + * "The Scientist and Engineer's Guide to DSP", Chapter 20. + * http://www.dspguide.com/ + * + * For type 2 and Chebyshev filters in general read + * http://en.wikipedia.org/wiki/Chebyshev_filter + * + * Transformation from lowpass to bandpass/bandreject: + * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandPassZ.htm + * http://docs.dewresearch.com/DspHelp/html/IDH_LinearSystems_LowpassToBandStopZ.htm + * + */ + +/** + * SECTION:element-audiochebyshevfreqband + * @short_description: Chebyshev band pass and band reject filter + * + * + * + * Attenuates all frequencies outside (bandpass) or inside (bandreject) of a frequency + * band. The number of poles and the ripple parameter control the rolloff. + * + * + * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. + * some frequencies in the passband will be amplified by that value. A higher ripple value will allow + * a faster rolloff. + * + * + * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will + * be at most this value. A lower ripple value will allow a faster rolloff. + * + * + * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. + * + * Example launch line + * + * + * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequenc=6000 poles=4 ! audioconvert ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqband mode=band-reject lower-frequency=1000 upper-frequency=4000 ripple=0.2 ! audioconvert ! alsasink + * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqband mode=band-pass lower-frequency=1000 upper-frequency=4000 type=2 ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include + +#include "audiochebyshevfreqband.h" + +#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_band_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioChebyshevFreqBand", + "Filter/Effect/Audio", + "Chebyshev band pass and band reject filter", + "Sebastian Dröge "); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_MODE, + PROP_TYPE, + PROP_LOWER_FREQUENCY, + PROP_UPPER_FREQUENCY, + PROP_RIPPLE, + PROP_POLES +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width = (int) { 32, 64 }, " \ + " endianness = (int) BYTE_ORDER," \ + " rate = (int) [ 1, MAX ]," \ + " channels = (int) [ 1, MAX ]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_band_debug, "audiochebyshevfreqband", 0, "audiochebyshevfreqband element"); + +GST_BOILERPLATE_FULL (GstAudioChebyshevFreqBand, gst_audio_chebyshev_freq_band, + GstAudioFilter, GST_TYPE_AUDIO_FILTER, DEBUG_INIT); + +static void gst_audio_chebyshev_freq_band_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_audio_chebyshev_freq_band_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_chebyshev_freq_band_setup (GstAudioFilter * filter, + GstRingBufferSpec * format); +static GstFlowReturn +gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_audio_chebyshev_freq_band_start (GstBaseTransform * base); + +static void process_64 (GstAudioChebyshevFreqBand * filter, + gdouble * data, guint num_samples); +static void process_32 (GstAudioChebyshevFreqBand * filter, + gfloat * data, guint num_samples); + +enum +{ + MODE_BAND_PASS = 0, + MODE_BAND_REJECT +}; + +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE (gst_audio_chebyshev_freq_band_mode_get_type ()) +static GType +gst_audio_chebyshev_freq_band_mode_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {MODE_BAND_PASS, "Band pass (default)", + "band-pass"}, + {MODE_BAND_REJECT, "Band reject", + "band-reject"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioChebyshevFreqBandMode", values); + } + return gtype; +} + +/* GObject vmethod implementations */ + +static void +gst_audio_chebyshev_freq_band_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details (element_class, &element_details); + + caps = gst_caps_from_string (ALLOWED_CAPS); + gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), + caps); + gst_caps_unref (caps); +} + +static void +gst_audio_chebyshev_freq_band_dispose (GObject * object) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_audio_chebyshev_freq_band_class_init (GstAudioChebyshevFreqBandClass * + klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + GstAudioFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_chebyshev_freq_band_set_property; + gobject_class->get_property = gst_audio_chebyshev_freq_band_get_property; + gobject_class->dispose = gst_audio_chebyshev_freq_band_dispose; + + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", + "Low pass or high pass mode", GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND_MODE, + MODE_BAND_PASS, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_TYPE, + g_param_spec_int ("type", "Type", + "Type of the chebychev filter", 1, 2, + 1, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_LOWER_FREQUENCY, + g_param_spec_float ("lower-frequency", "Lower frequency", + "Start frequency of the band (Hz)", 0.0, G_MAXFLOAT, + 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_UPPER_FREQUENCY, + g_param_spec_float ("upper-frequency", "Upper frequency", + "Stop frequency of the band (Hz)", 0.0, G_MAXFLOAT, + 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_RIPPLE, + g_param_spec_float ("ripple", "Ripple", + "Amount of ripple (dB)", 0.0, G_MAXFLOAT, + 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_POLES, + g_param_spec_int ("poles", "Poles", + "Number of poles to use, will be rounded up to the next multiply of four", + 4, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + filter_class->setup = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_setup); + trans_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_transform_ip); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_band_start); +} + +static void +gst_audio_chebyshev_freq_band_init (GstAudioChebyshevFreqBand * filter, + GstAudioChebyshevFreqBandClass * klass) +{ + filter->lower_frequency = filter->upper_frequency = 0.0; + filter->mode = MODE_BAND_PASS; + filter->type = 1; + filter->poles = 4; + filter->ripple = 0.25; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); + + filter->have_coeffs = FALSE; + filter->num_a = 0; + filter->num_b = 0; + filter->channels = NULL; +} + +static void +generate_biquad_coefficients (GstAudioChebyshevFreqBand * filter, + gint p, gdouble * a0, gdouble * a1, gdouble * a2, gdouble * a3, + gdouble * a4, gdouble * b1, gdouble * b2, gdouble * b3, gdouble * b4) +{ + gint np = filter->poles / 2; + gdouble ripple = filter->ripple; + + /* pole location in s-plane */ + gdouble rp, ip; + + /* zero location in s-plane */ + gdouble rz = 0.0, iz = 0.0; + + /* transfer function coefficients for the z-plane */ + gdouble x0, x1, x2, y1, y2; + gint type = filter->type; + + /* Calculate pole location for lowpass at frequency 1 */ + { + gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + + rp = -sin (angle); + ip = cos (angle); + } + + /* If we allow ripple, move the pole from the unit + * circle to an ellipse and keep cutoff at frequency 1 */ + if (ripple > 0 && type == 1) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + + vx = (1.0 / np) * asinh (1.0 / es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } else if (type == 2) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + vx = (1.0 / np) * asinh (es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } + + /* Calculate inverse of the pole location to move from + * type I to type II */ + if (type == 2) { + gdouble mag2 = rp * rp + ip * ip; + + rp /= mag2; + ip /= mag2; + } + + /* Calculate zero location for frequency 1 on the + * unit circle for type 2 */ + if (type == 2) { + gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble mag2; + + rz = 0.0; + iz = cos (angle); + mag2 = rz * rz + iz * iz; + rz /= mag2; + iz /= mag2; + } + + /* Convert from s-domain to z-domain by + * using the bilinear Z-transform, i.e. + * substitute s by (2/t)*((z-1)/(z+1)) + * with t = 2 * tan(0.5). + */ + if (type == 1) { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t) / d; + x1 = 2.0 * x0; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } else { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t * iz * iz + 4.0) / d; + x1 = (-8.0 + 2.0 * iz * iz * t * t) / d; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } + + /* Convert from lowpass at frequency 1 to either bandpass + * or band reject. + * + * For bandpass substitute z^(-1) with: + * + * -2 -1 + * -z + alpha * z - beta + * ---------------------------- + * -2 -1 + * beta * z - alpha * z + 1 + * + * alpha = (2*a*b)/(1+b) + * beta = (b-1)/(b+1) + * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) + * b = tan(1/2) * cot((w1 - w0)/2) + * + * For bandreject substitute z^(-1) with: + * + * -2 -1 + * z - alpha * z + beta + * ---------------------------- + * -2 -1 + * beta * z - alpha * z + 1 + * + * alpha = (2*a)/(1+b) + * beta = (1-b)/(1+b) + * a = cos((w1 + w0)/2) / cos((w1 - w0)/2) + * b = tan(1/2) * tan((w1 - w0)/2) + * + */ + { + gdouble a, b, d; + gdouble alpha, beta; + gdouble w0 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w1 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + + if (filter->mode == MODE_BAND_PASS) { + a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); + b = tan (1.0 / 2.0) / tan ((w1 - w0) / 2.0); + + alpha = (2.0 * a * b) / (1.0 + b); + beta = (b - 1.0) / (b + 1.0); + + d = 1.0 + beta * (y1 - beta * y2); + + *a0 = (x0 + beta * (-x1 + beta * x2)) / d; + *a1 = (alpha * (-2.0 * x0 + x1 + beta * x1 - 2.0 * beta * x2)) / d; + *a2 = + (-x1 - beta * beta * x1 + 2.0 * beta * (x0 + x2) + + alpha * alpha * (x0 - x1 + x2)) / d; + *a3 = (alpha * (x1 + beta * (-2.0 * x0 + x1) - 2.0 * x2)) / d; + *a4 = (beta * (beta * x0 - x1) + x2) / d; + *b1 = (alpha * (2.0 + y1 + beta * y1 - 2.0 * beta * y2)) / d; + *b2 = + (-y1 - beta * beta * y1 - alpha * alpha * (1.0 + y1 - y2) + + 2.0 * beta * (-1.0 + y2)) / d; + *b3 = (alpha * (y1 + beta * (2.0 + y1) - 2.0 * y2)) / d; + *b4 = (-beta * beta - beta * y1 + y2) / d; + } else { + a = cos ((w1 + w0) / 2.0) / cos ((w1 - w0) / 2.0); + b = tan (1.0 / 2.0) * tan ((w1 - w0) / 2.0); + + alpha = (2.0 * a) / (1.0 + b); + beta = (1.0 - b) / (1.0 + b); + + d = -1.0 + beta * (beta * y2 + y1); + + *a0 = (-x0 - beta * x1 - beta * beta * x2) / d; + *a1 = (alpha * (2.0 * x0 + x1 + beta * x1 + 2.0 * beta * x2)) / d; + *a2 = + (-x1 - beta * beta * x1 - 2.0 * beta * (x0 + x2) - + alpha * alpha * (x0 + x1 + x2)) / d; + *a3 = (alpha * (x1 + beta * (2.0 * x0 + x1) + 2.0 * x2)) / d; + *a4 = (-beta * beta * x0 - beta * x1 - x2) / d; + *b1 = (alpha * (-2.0 + y1 + beta * y1 + 2.0 * beta * y2)) / d; + *b2 = + -(y1 + beta * beta * y1 + 2.0 * beta * (-1.0 + y2) + + alpha * alpha * (-1.0 + y1 + y2)) / d; + *b3 = (alpha * (beta * (-2.0 + y1) + y1 + 2.0 * y2)) / d; + *b4 = -(-beta * beta + beta * y1 + y2) / d; + } + } +} + +/* Evaluate the transfer function that corresponds to the IIR + * coefficients at zr + zi*I and return the magnitude */ +static gdouble +calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr, + gdouble zi) +{ + gdouble sum_ar, sum_ai; + gdouble sum_br, sum_bi; + gdouble gain_r, gain_i; + + gdouble sum_r_old; + gdouble sum_i_old; + + gint i; + + sum_ar = 0.0; + sum_ai = 0.0; + for (i = num_a; i >= 0; i--) { + sum_r_old = sum_ar; + sum_i_old = sum_ai; + + sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; + sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; + } + + sum_br = 0.0; + sum_bi = 0.0; + for (i = num_b; i >= 0; i--) { + sum_r_old = sum_br; + sum_i_old = sum_bi; + + sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; + sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; + } + sum_br += 1.0; + sum_bi += 0.0; + + gain_r = + (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + gain_i = + (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + + return (sqrt (gain_r * gain_r + gain_i * gain_i)); +} + +static void +generate_coefficients (GstAudioChebyshevFreqBand * filter) +{ + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + if (GST_AUDIO_FILTER (filter)->format.rate == 0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + GST_LOG_OBJECT (filter, "rate was not set yet"); + return; + } + + filter->have_coeffs = TRUE; + + if (filter->upper_frequency <= filter->lower_frequency) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_BAND_PASS) ? 0.0 : 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + GST_LOG_OBJECT (filter, "frequency band had no or negative dimension"); + return; + } + + if (filter->upper_frequency > GST_AUDIO_FILTER (filter)->format.rate / 2) { + filter->upper_frequency = GST_AUDIO_FILTER (filter)->format.rate / 2; + GST_LOG_OBJECT (filter, "clipped upper frequency to nyquist frequency"); + } + + if (filter->lower_frequency < 0.0) { + filter->lower_frequency = 0.0; + GST_LOG_OBJECT (filter, "clipped lower frequency to 0.0"); + } + + /* Calculate coefficients for the chebyshev filter */ + { + gint np = filter->poles; + gdouble *a, *b; + gint i, p; + + filter->num_a = np + 1; + filter->a = a = g_new0 (gdouble, np + 5); + filter->num_b = np + 1; + filter->b = b = g_new0 (gdouble, np + 5); + + filter->channels = g_new0 (GstAudioChebyshevFreqBandChannelCtx, channels); + for (i = 0; i < channels; i++) { + GstAudioChebyshevFreqBandChannelCtx *ctx = &filter->channels[i]; + + ctx->x = g_new0 (gdouble, np + 1); + ctx->y = g_new0 (gdouble, np + 1); + } + + /* Calculate transfer function coefficients */ + a[4] = 1.0; + b[4] = 1.0; + + for (p = 1; p <= np / 4; p++) { + gdouble a0, a1, a2, a3, a4, b1, b2, b3, b4; + gdouble *ta = g_new0 (gdouble, np + 5); + gdouble *tb = g_new0 (gdouble, np + 5); + + generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &a3, &a4, &b1, + &b2, &b3, &b4); + + memcpy (ta, a, sizeof (gdouble) * (np + 5)); + memcpy (tb, b, sizeof (gdouble) * (np + 5)); + + /* add the new coefficients for the new two poles + * to the cascade by multiplication of the transfer + * functions */ + for (i = 4; i < np + 5; i++) { + a[i] = + a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2] + a3 * ta[i - 3] + + a4 * ta[i - 4]; + b[i] = + tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2] - b3 * tb[i - 3] - + b4 * tb[i - 4]; + } + g_free (ta); + g_free (tb); + } + + /* Move coefficients to the beginning of the array + * and multiply the b coefficients with -1 to move from + * the transfer function's coefficients to the difference + * equation's coefficients */ + b[4] = 0.0; + for (i = 0; i <= np; i++) { + a[i] = a[i + 4]; + b[i] = -b[i + 4]; + } + + /* Normalize to unity gain at frequency 0 and frequency + * 0.5 for bandreject and unity gain at band center frequency + * for bandpass */ + if (filter->mode == MODE_BAND_REJECT) { + /* gain is sqrt(H(0)*H(0.5)) */ + + gdouble gain1 = calculate_gain (a, b, np, np, 1.0, 0.0); + gdouble gain2 = calculate_gain (a, b, np, np, -1.0, 0.0); + + gain1 = sqrt (gain1 * gain2); + + for (i = 0; i <= np; i++) { + a[i] /= gain1; + } + } else { + /* gain is H(wc), wc = center frequency */ + + gdouble w1 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w2 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w0 = (w2 + w1) / 2.0; + gdouble zr = cos (w0), zi = sin (w0); + gdouble gain = calculate_gain (a, b, np, np, zr, zi); + + for (i = 0; i <= np; i++) { + a[i] /= gain; + } + } + + GST_LOG_OBJECT (filter, + "Generated IIR coefficients for the Chebyshev filter"); + GST_LOG_OBJECT (filter, + "mode: %s, type: %d, poles: %d, lower-frequency: %.2f Hz, upper-frequency: %.2f Hz, ripple: %.2f dB", + (filter->mode == MODE_BAND_PASS) ? "band-pass" : "band-reject", + filter->type, filter->poles, filter->lower_frequency, + filter->upper_frequency, filter->ripple); + + GST_LOG_OBJECT (filter, "%.2f dB gain @ 0Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0))); + { + gdouble w1 = + 2.0 * M_PI * (filter->lower_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w2 = + 2.0 * M_PI * (filter->upper_frequency / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble w0 = (w2 + w1) / 2.0; + gdouble zr, zi; + + zr = cos (w1); + zi = sin (w1); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->lower_frequency); + zr = cos (w0); + zi = sin (w0); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) ((filter->lower_frequency + filter->upper_frequency) / 2.0)); + zr = cos (w2); + zi = sin (w2); + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->upper_frequency); + } + GST_LOG_OBJECT (filter, "%.2f dB gain @ %dHz", + 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)), + GST_AUDIO_FILTER (filter)->format.rate / 2); + } +} + +static void +gst_audio_chebyshev_freq_band_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + switch (prop_id) { + case PROP_MODE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->mode = g_value_get_enum (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_TYPE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->type = g_value_get_int (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_LOWER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (filter); + filter->lower_frequency = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_UPPER_FREQUENCY: + GST_BASE_TRANSFORM_LOCK (filter); + filter->upper_frequency = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_RIPPLE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->ripple = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_POLES: + GST_BASE_TRANSFORM_LOCK (filter); + filter->poles = GST_ROUND_UP_4 (g_value_get_int (value)); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_chebyshev_freq_band_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (object); + + switch (prop_id) { + case PROP_MODE: + g_value_set_enum (value, filter->mode); + break; + case PROP_TYPE: + g_value_set_int (value, filter->type); + break; + case PROP_LOWER_FREQUENCY: + g_value_set_float (value, filter->lower_frequency); + break; + case PROP_UPPER_FREQUENCY: + g_value_set_float (value, filter->upper_frequency); + break; + case PROP_RIPPLE: + g_value_set_float (value, filter->ripple); + break; + case PROP_POLES: + g_value_set_int (value, filter->poles); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_chebyshev_freq_band_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + gboolean ret = TRUE; + + if (format->width == 32) + filter->process = (GstAudioChebyshevFreqBandProcessFunc) + process_32; + else if (format->width == 64) + filter->process = (GstAudioChebyshevFreqBandProcessFunc) + process_64; + else + ret = FALSE; + + filter->have_coeffs = FALSE; + + return ret; +} + +static inline gdouble +process (GstAudioChebyshevFreqBand * filter, + GstAudioChebyshevFreqBandChannelCtx * ctx, gdouble x0) +{ + gdouble val = filter->a[0] * x0; + gint i, j; + + for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) { + val += filter->a[i] * ctx->x[j]; + j--; + if (j < 0) + j = filter->num_a - 1; + } + + for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) { + val += filter->b[i] * ctx->y[j]; + j--; + if (j < 0) + j = filter->num_b - 1; + } + + if (ctx->x) { + ctx->x_pos++; + if (ctx->x_pos > filter->num_a - 1) + ctx->x_pos = 0; + ctx->x[ctx->x_pos] = x0; + } + + if (ctx->y) { + ctx->y_pos++; + if (ctx->y_pos > filter->num_b - 1) + ctx->y_pos = 0; + + ctx->y[ctx->y_pos] = val; + } + + return val; +} + +static void +process_64 (GstAudioChebyshevFreqBand * filter, + gdouble * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +static void +process_32 (GstAudioChebyshevFreqBand * filter, + gfloat * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_chebyshev_freq_band_transform_ip (GstBaseTransform * base, + GstBuffer * buf) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); + + if (!filter->have_coeffs) + generate_coefficients (filter); + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} + +static gboolean +gst_audio_chebyshev_freq_band_start (GstBaseTransform * base) +{ + GstAudioChebyshevFreqBand *filter = GST_AUDIO_CHEBYSHEV_FREQ_BAND (base); + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + GstAudioChebyshevFreqBandChannelCtx *ctx; + gint i; + + /* Reset the history of input and output values if + * already existing */ + if (channels && filter->channels) { + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + if (ctx->x) + memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble)); + if (ctx->y) + memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble)); + } + } + return TRUE; +} diff --git a/gst/audiofx/audiochebyshevfreqband.h b/gst/audiofx/audiochebyshevfreqband.h new file mode 100644 index 00000000..e8c58074 --- /dev/null +++ b/gst/audiofx/audiochebyshevfreqband.h @@ -0,0 +1,79 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ +#define __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND (gst_audio_chebyshev_freq_band_get_type()) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBand)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_BAND_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)) +#define GST_AUDIO_CHEBYSHEV_FREQ_BAND_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND,GstAudioChebyshevFreqBandClass)) +typedef struct _GstAudioChebyshevFreqBand GstAudioChebyshevFreqBand; +typedef struct _GstAudioChebyshevFreqBandClass GstAudioChebyshevFreqBandClass; + +typedef void (*GstAudioChebyshevFreqBandProcessFunc) (GstAudioChebyshevFreqBand *, guint8 *, guint); + +typedef struct +{ + gdouble *x; + gint x_pos; + gdouble *y; + gint y_pos; +} GstAudioChebyshevFreqBandChannelCtx; + +struct _GstAudioChebyshevFreqBand +{ + GstAudioFilter audiofilter; + + gint mode; + gint type; + gint poles; + gfloat lower_frequency; + gfloat upper_frequency; + gfloat ripple; + + /* < private > */ + GstAudioChebyshevFreqBandProcessFunc process; + + gboolean have_coeffs; + gdouble *a; + gint num_a; + gdouble *b; + gint num_b; + GstAudioChebyshevFreqBandChannelCtx *channels; +}; + +struct _GstAudioChebyshevFreqBandClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_chebyshev_freq_band_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_BAND_H__ */ diff --git a/gst/audiofx/audiochebyshevfreqlimit.c b/gst/audiofx/audiochebyshevfreqlimit.c new file mode 100644 index 00000000..872b277d --- /dev/null +++ b/gst/audiofx/audiochebyshevfreqlimit.c @@ -0,0 +1,816 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +/* + * Chebyshev type 1 filter design based on + * "The Scientist and Engineer's Guide to DSP", Chapter 20. + * http://www.dspguide.com/ + * + * For type 2 and Chebyshev filters in general read + * http://en.wikipedia.org/wiki/Chebyshev_filter + * + */ + +/** + * SECTION:element-audiochebyshevfreqlimit + * @short_description: Chebyshev low pass and high pass filter + * + * + * + * Attenuates all frequencies above the cutoff frequency (low-pass) or all frequencies below the + * cutoff frequency (high-pass). The number of poles and the ripple parameter control the rolloff. + * + * + * For type 1 the ripple parameter specifies how much ripple in dB is allowed in the passband, i.e. + * some frequencies in the passband will be amplified by that value. A higher ripple value will allow + * a faster rolloff. + * + * + * For type 2 the ripple parameter specifies the stopband attenuation. In the stopband the gain will + * be at most this value. A lower ripple value will allow a faster rolloff. + * + * + * As a special case, a Chebyshev type 1 filter with no ripple is a Butterworth filter. + * + * Example launch line + * + * + * gst-launch audiotestsrc freq=1500 ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=1000 poles=4 ! audioconvert ! alsasink + * gst-launch filesrc location="melo1.ogg" ! oggdemux ! vorbisdec ! audioconvert ! audiochebyshevfreqlimit mode=high-pass cutoff=400 ripple=0.2 ! audioconvert ! alsasink + * gst-launch audiotestsrc wave=white-noise ! audioconvert ! audiochebyshevfreqlimit mode=low-pass cutoff=800 type=2 ! audioconvert ! alsasink + * + * + * + */ + +#ifdef HAVE_CONFIG_H +#include "config.h" +#endif + +#include +#include +#include +#include +#include + +#include + +#include "audiochebyshevfreqlimit.h" + +#define GST_CAT_DEFAULT gst_audio_chebyshev_freq_limit_debug +GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); + +static const GstElementDetails element_details = +GST_ELEMENT_DETAILS ("AudioChebyshevFreqLimit", + "Filter/Effect/Audio", + "Chebyshev low pass and high pass filter", + "Sebastian Dröge "); + +/* Filter signals and args */ +enum +{ + /* FILL ME */ + LAST_SIGNAL +}; + +enum +{ + PROP_0, + PROP_MODE, + PROP_TYPE, + PROP_CUTOFF, + PROP_RIPPLE, + PROP_POLES +}; + +#define ALLOWED_CAPS \ + "audio/x-raw-float," \ + " width = (int) { 32, 64 }, " \ + " endianness = (int) BYTE_ORDER," \ + " rate = (int) [ 1, MAX ]," \ + " channels = (int) [ 1, MAX ]" + +#define DEBUG_INIT(bla) \ + GST_DEBUG_CATEGORY_INIT (gst_audio_chebyshev_freq_limit_debug, "audiochebyshevfreqlimit", 0, "audiochebyshevfreqlimit element"); + +GST_BOILERPLATE_FULL (GstAudioChebyshevFreqLimit, + gst_audio_chebyshev_freq_limit, GstAudioFilter, GST_TYPE_AUDIO_FILTER, + DEBUG_INIT); + +static void gst_audio_chebyshev_freq_limit_set_property (GObject * object, + guint prop_id, const GValue * value, GParamSpec * pspec); +static void gst_audio_chebyshev_freq_limit_get_property (GObject * object, + guint prop_id, GValue * value, GParamSpec * pspec); + +static gboolean gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * filter, + GstRingBufferSpec * format); +static GstFlowReturn +gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, + GstBuffer * buf); +static gboolean gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base); + +static void process_64 (GstAudioChebyshevFreqLimit * filter, + gdouble * data, guint num_samples); +static void process_32 (GstAudioChebyshevFreqLimit * filter, + gfloat * data, guint num_samples); + +enum +{ + MODE_LOW_PASS = 0, + MODE_HIGH_PASS +}; + +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE (gst_audio_chebyshev_freq_limit_mode_get_type ()) +static GType +gst_audio_chebyshev_freq_limit_mode_get_type (void) +{ + static GType gtype = 0; + + if (gtype == 0) { + static const GEnumValue values[] = { + {MODE_LOW_PASS, "Low pass (default)", + "low-pass"}, + {MODE_HIGH_PASS, "High pass", + "high-pass"}, + {0, NULL, NULL} + }; + + gtype = g_enum_register_static ("GstAudioChebyshevFreqLimitMode", values); + } + return gtype; +} + +/* GObject vmethod implementations */ + +static void +gst_audio_chebyshev_freq_limit_base_init (gpointer klass) +{ + GstElementClass *element_class = GST_ELEMENT_CLASS (klass); + GstCaps *caps; + + gst_element_class_set_details (element_class, &element_details); + + caps = gst_caps_from_string (ALLOWED_CAPS); + gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), + caps); + gst_caps_unref (caps); +} + +static void +gst_audio_chebyshev_freq_limit_dispose (GObject * object) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i, channels = GST_AUDIO_FILTER (filter)->format.channels; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + G_OBJECT_CLASS (parent_class)->dispose (object); +} + +static void +gst_audio_chebyshev_freq_limit_class_init (GstAudioChebyshevFreqLimitClass * + klass) +{ + GObjectClass *gobject_class; + GstBaseTransformClass *trans_class; + GstAudioFilterClass *filter_class; + + gobject_class = (GObjectClass *) klass; + trans_class = (GstBaseTransformClass *) klass; + filter_class = (GstAudioFilterClass *) klass; + + gobject_class->set_property = gst_audio_chebyshev_freq_limit_set_property; + gobject_class->get_property = gst_audio_chebyshev_freq_limit_get_property; + gobject_class->dispose = gst_audio_chebyshev_freq_limit_dispose; + + g_object_class_install_property (gobject_class, PROP_MODE, + g_param_spec_enum ("mode", "Mode", + "Low pass or high pass mode", + GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT_MODE, MODE_LOW_PASS, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_TYPE, + g_param_spec_int ("type", "Type", "Type of the chebychev filter", 1, 2, 1, + G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_CUTOFF, + g_param_spec_float ("cutoff", "Cutoff", "Cut off frequency (Hz)", 0.0, + G_MAXFLOAT, 0.0, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_RIPPLE, + g_param_spec_float ("ripple", "Ripple", "Amount of ripple (dB)", 0.0, + G_MAXFLOAT, 0.25, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + g_object_class_install_property (gobject_class, PROP_POLES, + g_param_spec_int ("poles", "Poles", + "Number of poles to use, will be rounded up to the next even number", + 2, 32, 4, G_PARAM_READWRITE | GST_PARAM_CONTROLLABLE)); + + filter_class->setup = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_setup); + trans_class->transform_ip = + GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_transform_ip); + trans_class->start = GST_DEBUG_FUNCPTR (gst_audio_chebyshev_freq_limit_start); +} + +static void +gst_audio_chebyshev_freq_limit_init (GstAudioChebyshevFreqLimit * filter, + GstAudioChebyshevFreqLimitClass * klass) +{ + filter->cutoff = 0.0; + filter->mode = MODE_LOW_PASS; + filter->type = 1; + filter->poles = 4; + filter->ripple = 0.25; + gst_base_transform_set_in_place (GST_BASE_TRANSFORM (filter), TRUE); + + filter->have_coeffs = FALSE; + filter->num_a = 0; + filter->num_b = 0; + filter->channels = NULL; +} + +static void +generate_biquad_coefficients (GstAudioChebyshevFreqLimit * filter, + gint p, gdouble * a0, gdouble * a1, gdouble * a2, + gdouble * b1, gdouble * b2) +{ + gint np = filter->poles; + gdouble ripple = filter->ripple; + + /* pole location in s-plane */ + gdouble rp, ip; + + /* zero location in s-plane */ + gdouble rz = 0.0, iz = 0.0; + + /* transfer function coefficients for the z-plane */ + gdouble x0, x1, x2, y1, y2; + gint type = filter->type; + + /* Calculate pole location for lowpass at frequency 1 */ + { + gdouble angle = (M_PI / 2.0) * (2.0 * p - 1) / np; + + rp = -sin (angle); + ip = cos (angle); + } + + /* If we allow ripple, move the pole from the unit + * circle to an ellipse and keep cutoff at frequency 1 */ + if (ripple > 0 && type == 1) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + + vx = (1.0 / np) * asinh (1.0 / es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } else if (type == 2) { + gdouble es, vx; + + es = sqrt (pow (10.0, ripple / 10.0) - 1.0); + vx = (1.0 / np) * asinh (es); + rp = rp * sinh (vx); + ip = ip * cosh (vx); + } + + /* Calculate inverse of the pole location to convert from + * type I to type II */ + if (type == 2) { + gdouble mag2 = rp * rp + ip * ip; + + rp /= mag2; + ip /= mag2; + } + + /* Calculate zero location for frequency 1 on the + * unit circle for type 2 */ + if (type == 2) { + gdouble angle = M_PI / (np * 2.0) + ((p - 1) * M_PI) / (np); + gdouble mag2; + + rz = 0.0; + iz = cos (angle); + mag2 = rz * rz + iz * iz; + rz /= mag2; + iz /= mag2; + } + + /* Convert from s-domain to z-domain by + * using the bilinear Z-transform, i.e. + * substitute s by (2/t)*((z-1)/(z+1)) + * with t = 2 * tan(0.5). + */ + if (type == 1) { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t) / d; + x1 = 2.0 * x0; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } else { + gdouble t, m, d; + + t = 2.0 * tan (0.5); + m = rp * rp + ip * ip; + d = 4.0 - 4.0 * rp * t + m * t * t; + + x0 = (t * t * iz * iz + 4.0) / d; + x1 = (-8.0 + 2.0 * iz * iz * t * t) / d; + x2 = x0; + y1 = (8.0 - 2.0 * m * t * t) / d; + y2 = (-4.0 - 4.0 * rp * t - m * t * t) / d; + } + + /* Convert from lowpass at frequency 1 to either lowpass + * or highpass. + * + * For lowpass substitute z^(-1) with: + * -1 + * z - k + * ------------ + * -1 + * 1 - k * z + * + * k = sin((1-w)/2) / sin((1+w)/2) + * + * For highpass substitute z^(-1) with: + * + * -1 + * -z - k + * ------------ + * -1 + * 1 + k * z + * + * k = -cos((1+w)/2) / cos((1-w)/2) + * + */ + { + gdouble k, d; + gdouble omega = + 2.0 * M_PI * (filter->cutoff / GST_AUDIO_FILTER (filter)->format.rate); + + if (filter->mode == MODE_LOW_PASS) + k = sin ((1.0 - omega) / 2.0) / sin ((1.0 + omega) / 2.0); + else + k = -cos ((omega + 1.0) / 2.0) / cos ((omega - 1.0) / 2.0); + + d = 1.0 + y1 * k - y2 * k * k; + *a0 = (x0 + k * (-x1 + k * x2)) / d; + *a1 = (x1 + k * k * x1 - 2.0 * k * (x0 + x2)) / d; + *a2 = (x0 * k * k - x1 * k + x2) / d; + *b1 = (2.0 * k + y1 + y1 * k * k - 2.0 * y2 * k) / d; + *b2 = (-k * k - y1 * k + y2) / d; + + if (filter->mode == MODE_HIGH_PASS) { + *a1 = -*a1; + *b1 = -*b1; + } + } +} + +/* Evaluate the transfer function that corresponds to the IIR + * coefficients at zr + zi*I and return the magnitude */ +static gdouble +calculate_gain (gdouble * a, gdouble * b, gint num_a, gint num_b, gdouble zr, + gdouble zi) +{ + gdouble sum_ar, sum_ai; + gdouble sum_br, sum_bi; + gdouble gain_r, gain_i; + + gdouble sum_r_old; + gdouble sum_i_old; + + gint i; + + sum_ar = 0.0; + sum_ai = 0.0; + for (i = num_a; i >= 0; i--) { + sum_r_old = sum_ar; + sum_i_old = sum_ai; + + sum_ar = (sum_r_old * zr - sum_i_old * zi) + a[i]; + sum_ai = (sum_r_old * zi + sum_i_old * zr) + 0.0; + } + + sum_br = 0.0; + sum_bi = 0.0; + for (i = num_b; i >= 0; i--) { + sum_r_old = sum_br; + sum_i_old = sum_bi; + + sum_br = (sum_r_old * zr - sum_i_old * zi) - b[i]; + sum_bi = (sum_r_old * zi + sum_i_old * zr) - 0.0; + } + sum_br += 1.0; + sum_bi += 0.0; + + gain_r = + (sum_ar * sum_br + sum_ai * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + gain_i = + (sum_ai * sum_br - sum_ar * sum_bi) / (sum_br * sum_br + sum_bi * sum_bi); + + return (sqrt (gain_r * gain_r + gain_i * gain_i)); +} + +static void +generate_coefficients (GstAudioChebyshevFreqLimit * filter) +{ + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + + if (filter->a) { + g_free (filter->a); + filter->a = NULL; + } + + if (filter->b) { + g_free (filter->b); + filter->b = NULL; + } + + if (filter->channels) { + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i; + + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + g_free (ctx->x); + g_free (ctx->y); + } + + g_free (filter->channels); + filter->channels = NULL; + } + + if (GST_AUDIO_FILTER (filter)->format.rate == 0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "rate was not set yet"); + return; + } + + filter->have_coeffs = TRUE; + + if (filter->cutoff >= GST_AUDIO_FILTER (filter)->format.rate / 2.0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 1.0 : 0.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "cutoff was higher than nyquist frequency"); + return; + } else if (filter->cutoff <= 0.0) { + filter->num_a = 1; + filter->a = g_new0 (gdouble, 1); + filter->a[0] = (filter->mode == MODE_LOW_PASS) ? 0.0 : 1.0; + filter->num_b = 0; + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + GST_LOG_OBJECT (filter, "cutoff is lower than zero"); + return; + } + + /* Calculate coefficients for the chebyshev filter */ + { + gint np = filter->poles; + gdouble *a, *b; + gint i, p; + + filter->num_a = np + 1; + filter->a = a = g_new0 (gdouble, np + 3); + filter->num_b = np + 1; + filter->b = b = g_new0 (gdouble, np + 3); + + filter->channels = g_new0 (GstAudioChebyshevFreqLimitChannelCtx, channels); + for (i = 0; i < channels; i++) { + GstAudioChebyshevFreqLimitChannelCtx *ctx = &filter->channels[i]; + + ctx->x = g_new0 (gdouble, np + 1); + ctx->y = g_new0 (gdouble, np + 1); + } + + /* Calculate transfer function coefficients */ + a[2] = 1.0; + b[2] = 1.0; + + for (p = 1; p <= np / 2; p++) { + gdouble a0, a1, a2, b1, b2; + gdouble *ta = g_new0 (gdouble, np + 3); + gdouble *tb = g_new0 (gdouble, np + 3); + + generate_biquad_coefficients (filter, p, &a0, &a1, &a2, &b1, &b2); + + memcpy (ta, a, sizeof (gdouble) * (np + 3)); + memcpy (tb, b, sizeof (gdouble) * (np + 3)); + + /* add the new coefficients for the new two poles + * to the cascade by multiplication of the transfer + * functions */ + for (i = 2; i < np + 3; i++) { + a[i] = a0 * ta[i] + a1 * ta[i - 1] + a2 * ta[i - 2]; + b[i] = tb[i] - b1 * tb[i - 1] - b2 * tb[i - 2]; + } + g_free (ta); + g_free (tb); + } + + /* Move coefficients to the beginning of the array + * and multiply the b coefficients with -1 to move from + * the transfer function's coefficients to the difference + * equation's coefficients */ + b[2] = 0.0; + for (i = 0; i <= np; i++) { + a[i] = a[i + 2]; + b[i] = -b[i + 2]; + } + + /* Normalize to unity gain at frequency 0 for lowpass + * and frequency 0.5 for highpass */ + { + gdouble gain; + + if (filter->mode == MODE_LOW_PASS) + gain = calculate_gain (a, b, np, np, 1.0, 0.0); + else + gain = calculate_gain (a, b, np, np, -1.0, 0.0); + + for (i = 0; i <= np; i++) { + a[i] /= gain; + } + } + + GST_LOG_OBJECT (filter, + "Generated IIR coefficients for the Chebyshev filter"); + GST_LOG_OBJECT (filter, + "mode: %s, type: %d, poles: %d, cutoff: %.2f Hz, ripple: %.2f dB", + (filter->mode == MODE_LOW_PASS) ? "low-pass" : "high-pass", + filter->type, filter->poles, filter->cutoff, filter->ripple); + GST_LOG_OBJECT (filter, "%.2f dB gain @ 0 Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, 1.0, 0.0))); + { + gdouble wc = + 2.0 * M_PI * (filter->cutoff / + GST_AUDIO_FILTER (filter)->format.rate); + gdouble zr = cos (wc), zi = sin (wc); + + GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, zr, zi)), + (int) filter->cutoff); + } + GST_LOG_OBJECT (filter, "%.2f dB gain @ %d Hz", + 20.0 * log10 (calculate_gain (a, b, np, np, -1.0, 0.0)), + GST_AUDIO_FILTER (filter)->format.rate / 2); + } +} + +static void +gst_audio_chebyshev_freq_limit_set_property (GObject * object, guint prop_id, + const GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + switch (prop_id) { + case PROP_MODE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->mode = g_value_get_enum (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_TYPE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->type = g_value_get_int (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_CUTOFF: + GST_BASE_TRANSFORM_LOCK (filter); + filter->cutoff = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_RIPPLE: + GST_BASE_TRANSFORM_LOCK (filter); + filter->ripple = g_value_get_float (value); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + case PROP_POLES: + GST_BASE_TRANSFORM_LOCK (filter); + filter->poles = GST_ROUND_UP_2 (g_value_get_int (value)); + generate_coefficients (filter); + GST_BASE_TRANSFORM_UNLOCK (filter); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +static void +gst_audio_chebyshev_freq_limit_get_property (GObject * object, guint prop_id, + GValue * value, GParamSpec * pspec) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (object); + + switch (prop_id) { + case PROP_MODE: + g_value_set_enum (value, filter->mode); + break; + case PROP_TYPE: + g_value_set_int (value, filter->type); + break; + case PROP_CUTOFF: + g_value_set_float (value, filter->cutoff); + break; + case PROP_RIPPLE: + g_value_set_float (value, filter->ripple); + break; + case PROP_POLES: + g_value_set_int (value, filter->poles); + break; + default: + G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); + break; + } +} + +/* GstAudioFilter vmethod implementations */ + +static gboolean +gst_audio_chebyshev_freq_limit_setup (GstAudioFilter * base, + GstRingBufferSpec * format) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + gboolean ret = TRUE; + + if (format->width == 32) + filter->process = (GstAudioChebyshevFreqLimitProcessFunc) + process_32; + else if (format->width == 64) + filter->process = (GstAudioChebyshevFreqLimitProcessFunc) + process_64; + else + ret = FALSE; + + filter->have_coeffs = FALSE; + + return ret; +} + +static inline gdouble +process (GstAudioChebyshevFreqLimit * filter, + GstAudioChebyshevFreqLimitChannelCtx * ctx, gdouble x0) +{ + gdouble val = filter->a[0] * x0; + gint i, j; + + for (i = 1, j = ctx->x_pos; i < filter->num_a; i++) { + val += filter->a[i] * ctx->x[j]; + j--; + if (j < 0) + j = filter->num_a - 1; + } + + for (i = 1, j = ctx->y_pos; i < filter->num_b; i++) { + val += filter->b[i] * ctx->y[j]; + j--; + if (j < 0) + j = filter->num_b - 1; + } + + if (ctx->x) { + ctx->x_pos++; + if (ctx->x_pos > filter->num_a - 1) + ctx->x_pos = 0; + ctx->x[ctx->x_pos] = x0; + } + + if (ctx->y) { + ctx->y_pos++; + if (ctx->y_pos > filter->num_b - 1) + ctx->y_pos = 0; + + ctx->y[ctx->y_pos] = val; + } + + return val; +} + +static void +process_64 (GstAudioChebyshevFreqLimit * filter, + gdouble * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +static void +process_32 (GstAudioChebyshevFreqLimit * filter, + gfloat * data, guint num_samples) +{ + gint i, j, channels = GST_AUDIO_FILTER (filter)->format.channels; + gdouble val; + + for (i = 0; i < num_samples / channels; i++) { + for (j = 0; j < channels; j++) { + val = process (filter, &filter->channels[j], *data); + *data++ = val; + } + } +} + +/* GstBaseTransform vmethod implementations */ +static GstFlowReturn +gst_audio_chebyshev_freq_limit_transform_ip (GstBaseTransform * base, + GstBuffer * buf) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + guint num_samples = + GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (filter)->format.width / 8); + + if (!gst_buffer_is_writable (buf)) + return GST_FLOW_OK; + + if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) + gst_object_sync_values (G_OBJECT (filter), GST_BUFFER_TIMESTAMP (buf)); + + if (!filter->have_coeffs) + generate_coefficients (filter); + + filter->process (filter, GST_BUFFER_DATA (buf), num_samples); + + return GST_FLOW_OK; +} + + +static gboolean +gst_audio_chebyshev_freq_limit_start (GstBaseTransform * base) +{ + GstAudioChebyshevFreqLimit *filter = GST_AUDIO_CHEBYSHEV_FREQ_LIMIT (base); + gint channels = GST_AUDIO_FILTER (filter)->format.channels; + GstAudioChebyshevFreqLimitChannelCtx *ctx; + gint i; + + /* Reset the history of input and output values if + * already existing */ + if (channels && filter->channels) { + for (i = 0; i < channels; i++) { + ctx = &filter->channels[i]; + if (ctx->x) + memset (ctx->x, 0, (filter->poles + 1) * sizeof (gdouble)); + if (ctx->y) + memset (ctx->y, 0, (filter->poles + 1) * sizeof (gdouble)); + } + } + return TRUE; +} diff --git a/gst/audiofx/audiochebyshevfreqlimit.h b/gst/audiofx/audiochebyshevfreqlimit.h new file mode 100644 index 00000000..4c87ba8e --- /dev/null +++ b/gst/audiofx/audiochebyshevfreqlimit.h @@ -0,0 +1,78 @@ +/* + * GStreamer + * Copyright (C) 2007 Sebastian Dröge + * + * This library is free software; you can redistribute it and/or + * modify it under the terms of the GNU Library General Public + * License as published by the Free Software Foundation; either + * version 2 of the License, or (at your option) any later version. + * + * This library is distributed in the hope that it will be useful, + * but WITHOUT ANY WARRANTY; without even the implied warranty of + * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU + * Library General Public License for more details. + * + * You should have received a copy of the GNU Library General Public + * License along with this library; if not, write to the + * Free Software Foundation, Inc., 59 Temple Place - Suite 330, + * Boston, MA 02111-1307, USA. + */ + +#ifndef __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ +#define __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ + +#include +#include +#include +#include + +G_BEGIN_DECLS +#define GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT (gst_audio_chebyshev_freq_limit_get_type()) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimit)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT(obj) (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_CAST((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) +#define GST_IS_AUDIO_CHEBYSHEV_FREQ_LIMIT_CLASS(klass) (G_TYPE_CHECK_CLASS_TYPE((klass) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT)) +#define GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_GET_CLASS(obj) (G_TYPE_INSTANCE_GET_CLASS((obj) ,GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT,GstAudioChebyshevFreqLimitClass)) +typedef struct _GstAudioChebyshevFreqLimit GstAudioChebyshevFreqLimit; +typedef struct _GstAudioChebyshevFreqLimitClass GstAudioChebyshevFreqLimitClass; + +typedef void (*GstAudioChebyshevFreqLimitProcessFunc) (GstAudioChebyshevFreqLimit *, guint8 *, guint); + +typedef struct +{ + gdouble *x; + gint x_pos; + gdouble *y; + gint y_pos; +} GstAudioChebyshevFreqLimitChannelCtx; + +struct _GstAudioChebyshevFreqLimit +{ + GstAudioFilter audiofilter; + + gint mode; + gint type; + gint poles; + gfloat cutoff; + gfloat ripple; + + /* < private > */ + GstAudioChebyshevFreqLimitProcessFunc process; + + gboolean have_coeffs; + gdouble *a; + gint num_a; + gdouble *b; + gint num_b; + GstAudioChebyshevFreqLimitChannelCtx *channels; +}; + +struct _GstAudioChebyshevFreqLimitClass +{ + GstAudioFilterClass parent; +}; + +GType gst_audio_chebyshev_freq_limit_get_type (void); + +G_END_DECLS +#endif /* __GST_AUDIO_CHEBYSHEV_FREQ_LIMIT_H__ */ diff --git a/gst/audiofx/audiofx.c b/gst/audiofx/audiofx.c index e6d84d24..2c198f32 100644 --- a/gst/audiofx/audiofx.c +++ b/gst/audiofx/audiofx.c @@ -29,6 +29,8 @@ #include "audioinvert.h" #include "audioamplify.h" #include "audiodynamic.h" +#include "audiochebyshevfreqlimit.h" +#include "audiochebyshevfreqband.h" /* entry point to initialize the plug-in * initialize the plug-in itself @@ -48,7 +50,11 @@ plugin_init (GstPlugin * plugin) gst_element_register (plugin, "audioamplify", GST_RANK_NONE, GST_TYPE_AUDIO_AMPLIFY) && gst_element_register (plugin, "audiodynamic", GST_RANK_NONE, - GST_TYPE_AUDIO_DYNAMIC)); + GST_TYPE_AUDIO_DYNAMIC) && + gst_element_register (plugin, "audiochebyshevfreqlimit", GST_RANK_NONE, + GST_TYPE_AUDIO_CHEBYSHEV_FREQ_LIMIT) && + gst_element_register (plugin, "audiochebyshevfreqband", GST_RANK_NONE, + GST_TYPE_AUDIO_CHEBYSHEV_FREQ_BAND)); } GST_PLUGIN_DEFINE (GST_VERSION_MAJOR, -- cgit