GStreamer Good Plugins 0.10 Plugins Reference Manual | ||||
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Top | Description | Object Hierarchy | Properties | Signals |
GObject +----GstObject +----GstElement +----GstBaseTransform +----GstAudioFilter +----GstAudioFXBaseIIRFilter +----GstAudioIIRFilter
audioiirfilter implements a generic audio IIR filter. Before usage the "a" and "b" properties have to be set to the filter coefficients that should be used.
The filter coefficients describe the numerator and denominator of the transfer function.
To change the filter coefficients whenever the sampling rate changes the "rate-changed" signal can be used. This should be done for most IIR filters as they're depending on the sampling rate.
/* GStreamer * Copyright (C) 2009 Sebastian Droege <sebastian.droege@collabora.co.uk> * * 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. */ /* This small sample application creates a lowpass IIR filter * and applies it to white noise. * See http://www.dspguide.com/ch19/2.htm for a description * of the IIR filter that is used. */ #include <string.h> #include <math.h> #include <gst/gst.h> /* Cutoff of 4000 Hz */ #define CUTOFF (4000.0) static gboolean on_message (GstBus * bus, GstMessage * message, gpointer user_data) { GMainLoop *loop = (GMainLoop *) user_data; switch (GST_MESSAGE_TYPE (message)) { case GST_MESSAGE_ERROR: g_error ("Got ERROR"); g_main_loop_quit (loop); break; case GST_MESSAGE_WARNING: g_warning ("Got WARNING"); g_main_loop_quit (loop); break; case GST_MESSAGE_EOS: g_main_loop_quit (loop); break; default: break; } return TRUE; } static void on_rate_changed (GstElement * element, gint rate, gpointer user_data) { GValueArray *va; GValue v = { 0, }; gdouble x; if (rate / 2.0 > CUTOFF) x = exp (-2.0 * M_PI * (CUTOFF / rate)); else x = 0.0; va = g_value_array_new (1); g_value_init (&v, G_TYPE_DOUBLE); g_value_set_double (&v, 1.0 - x); g_value_array_append (va, &v); g_value_reset (&v); g_object_set (G_OBJECT (element), "a", va, NULL); g_value_array_free (va); va = g_value_array_new (1); g_value_set_double (&v, x); g_value_array_append (va, &v); g_value_reset (&v); g_object_set (G_OBJECT (element), "b", va, NULL); g_value_array_free (va); } gint main (gint argc, gchar * argv[]) { GstElement *pipeline, *src, *filter, *conv, *sink; GstBus *bus; GMainLoop *loop; gst_init (NULL, NULL); pipeline = gst_element_factory_make ("pipeline", NULL); src = gst_element_factory_make ("audiotestsrc", NULL); g_object_set (G_OBJECT (src), "wave", 5, NULL); filter = gst_element_factory_make ("audioiirfilter", NULL); g_signal_connect (G_OBJECT (filter), "rate-changed", G_CALLBACK (on_rate_changed), NULL); conv = gst_element_factory_make ("audioconvert", NULL); sink = gst_element_factory_make ("autoaudiosink", NULL); g_return_val_if_fail (sink != NULL, -1); gst_bin_add_many (GST_BIN (pipeline), src, filter, conv, sink, NULL); if (!gst_element_link_many (src, filter, conv, sink, NULL)) { g_error ("Failed to link elements"); return -2; } loop = g_main_loop_new (NULL, FALSE); bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline)); gst_bus_add_signal_watch (bus); g_signal_connect (G_OBJECT (bus), "message", G_CALLBACK (on_message), loop); gst_object_unref (GST_OBJECT (bus)); if (gst_element_set_state (pipeline, GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) { g_error ("Failed to go into PLAYING state"); return -3; } g_main_loop_run (loop); gst_element_set_state (pipeline, GST_STATE_NULL); g_main_loop_unref (loop); gst_object_unref (pipeline); return 0; }
"a"
property "a" GValueArray* : Read / Write
Filter coefficients (numerator of transfer function).
"rate-changed"
signalvoid user_function (GstAudioIIRFilter *filter, gint rate, gpointer user_data) : Run Last
Will be emitted when the sampling rate changes. The callbacks will be called from the streaming thread and processing will stop until the event is handled.
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the filter on which the signal is emitted |
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the new sampling rate |
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user data set when the signal handler was connected. |