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Original commit message from CVS: * configure.ac: * gst/audiofx/Makefile.am: * gst/audiofx/audiofirfilter.c: (gst_audio_fir_filter_base_init), (gst_audio_fir_filter_class_init), (gst_audio_fir_filter_update_kernel), (gst_audio_fir_filter_init), (gst_audio_fir_filter_setup), (gst_audio_fir_filter_finalize), (gst_audio_fir_filter_set_property), (gst_audio_fir_filter_get_property): * gst/audiofx/audiofirfilter.h: * gst/audiofx/audiofx.c: (plugin_init): * gst/audiofx/audioiirfilter.c: (gst_audio_iir_filter_base_init), (gst_audio_iir_filter_class_init), (gst_audio_iir_filter_update_coefficients), (gst_audio_iir_filter_init), (gst_audio_iir_filter_setup), (gst_audio_iir_filter_finalize), (gst_audio_iir_filter_set_property), (gst_audio_iir_filter_get_property): * gst/audiofx/audioiirfilter.h: Add audioiirfilter and audiofirfilter elements which allow generic IIR/FIR filters to be implemented by providing the filter coefficients. Fixes bug #567577. * 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/gst-plugins-good-plugins.hierarchy: * docs/plugins/gst-plugins-good-plugins.signals: * docs/plugins/inspect/plugin-alaw.xml: * docs/plugins/inspect/plugin-audiofx.xml: * docs/plugins/inspect/plugin-avi.xml: * docs/plugins/inspect/plugin-flac.xml: * docs/plugins/inspect/plugin-mulaw.xml: * docs/plugins/inspect/plugin-video4linux2.xml: * docs/plugins/inspect/plugin-wavparse.xml: Add documentation for the audioiirfilter and audiofirfilter elements. * tests/check/Makefile.am: * tests/check/elements/audiofirfilter.c: (on_message), (on_rate_changed), (on_handoff), (GST_START_TEST), (audiofirfilter_suite): * tests/check/elements/audioiirfilter.c: (on_message), (on_rate_changed), (on_handoff), (GST_START_TEST), (audioiirfilter_suite): * tests/examples/Makefile.am: * tests/examples/audiofx/Makefile.am: * tests/examples/audiofx/firfilter-example.c: (on_message), (on_rate_changed), (main): * tests/examples/audiofx/iirfilter-example.c: (on_message), (on_rate_changed), (main): Add unit tests and example applications for the two filter elements.
161 lines
4.5 KiB
C
161 lines
4.5 KiB
C
/* GStreamer
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* Copyright (C) 2009 Sebastian Droege <sebastian.droege@collabora.co.uk>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 59 Temple Place - Suite 330,
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* Boston, MA 02111-1307, USA.
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*/
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/* This small sample application creates a bandpass FIR filter
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* by transforming the frequency response to the filter kernel.
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*/
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#include <string.h>
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#include <math.h>
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#include <gst/gst.h>
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#include <gst/fft/gstfftf64.h>
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static gboolean
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on_message (GstBus * bus, GstMessage * message, gpointer user_data)
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{
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GMainLoop *loop = (GMainLoop *) user_data;
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switch (GST_MESSAGE_TYPE (message)) {
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case GST_MESSAGE_ERROR:
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g_error ("Got ERROR");
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g_main_loop_quit (loop);
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break;
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case GST_MESSAGE_WARNING:
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g_warning ("Got WARNING");
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g_main_loop_quit (loop);
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break;
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case GST_MESSAGE_EOS:
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g_main_loop_quit (loop);
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break;
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default:
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break;
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}
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return TRUE;
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}
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static void
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on_rate_changed (GstElement * element, gint rate, gpointer user_data)
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{
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GValueArray *va;
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GValue v = { 0, };
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GstFFTF64 *fft;
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GstFFTF64Complex frequency_response[17];
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gdouble tmp[32];
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gdouble filter_kernel[32];
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guint i;
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/* Create the frequency response: zero outside
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* a small frequency band */
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for (i = 0; i < 17; i++) {
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if (i < 5 || i > 11)
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frequency_response[i].r = 0.0;
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else
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frequency_response[i].r = 1.0;
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frequency_response[i].i = 0.0;
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}
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/* Calculate the inverse FT of the frequency response */
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fft = gst_fft_f64_new (32, TRUE);
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gst_fft_f64_inverse_fft (fft, frequency_response, tmp);
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gst_fft_f64_free (fft);
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/* Shift the inverse FT of the frequency response by 16,
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* i.e. the half of the kernel length to get the
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* impulse response. See http://www.dspguide.com/ch17/1.htm
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* for more information.
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*/
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for (i = 0; i < 32; i++)
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filter_kernel[i] = tmp[(i + 16) % 32];
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/* Apply the hamming window to the impulse response to get
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* a better result than given from the rectangular window
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*/
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for (i = 0; i < 32; i++)
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filter_kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / 32));
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va = g_value_array_new (1);
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g_value_init (&v, G_TYPE_DOUBLE);
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for (i = 0; i < 32; i++) {
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g_value_set_double (&v, filter_kernel[i]);
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g_value_array_append (va, &v);
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g_value_reset (&v);
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}
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g_object_set (G_OBJECT (element), "kernel", va, NULL);
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/* Latency is 1/2 of the kernel length for this method of
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* calculating a filter kernel from the frequency response
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*/
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g_object_set (G_OBJECT (element), "latency", 32 / 2, NULL);
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g_value_array_free (va);
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}
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gint
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main (gint argc, gchar * argv[])
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{
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GstElement *pipeline, *src, *filter, *conv, *sink;
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GstBus *bus;
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GMainLoop *loop;
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gst_init (NULL, NULL);
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pipeline = gst_element_factory_make ("pipeline", NULL);
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src = gst_element_factory_make ("audiotestsrc", NULL);
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g_object_set (G_OBJECT (src), "wave", 5, NULL);
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filter = gst_element_factory_make ("audiofirfilter", NULL);
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g_signal_connect (G_OBJECT (filter), "rate-changed",
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G_CALLBACK (on_rate_changed), NULL);
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conv = gst_element_factory_make ("audioconvert", NULL);
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sink = gst_element_factory_make ("autoaudiosink", NULL);
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g_return_val_if_fail (sink != NULL, -1);
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gst_bin_add_many (GST_BIN (pipeline), src, filter, conv, sink, NULL);
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if (!gst_element_link_many (src, filter, conv, sink, NULL)) {
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g_error ("Failed to link elements");
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return -2;
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}
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loop = g_main_loop_new (NULL, FALSE);
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bus = gst_pipeline_get_bus (GST_PIPELINE (pipeline));
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gst_bus_add_signal_watch (bus);
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g_signal_connect (G_OBJECT (bus), "message", G_CALLBACK (on_message), loop);
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gst_object_unref (GST_OBJECT (bus));
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if (gst_element_set_state (pipeline,
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GST_STATE_PLAYING) == GST_STATE_CHANGE_FAILURE) {
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g_error ("Failed to go into PLAYING state");
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return -3;
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}
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g_main_loop_run (loop);
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gst_element_set_state (pipeline, GST_STATE_NULL);
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g_main_loop_unref (loop);
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gst_object_unref (pipeline);
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return 0;
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}
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