Add audioiirfilter and audiofirfilter elements which allow generic IIR/FIR filters to be implemented by providing the...

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.

4 files changed, 307 insertions, 2 deletions

diff --git a/tests/examples/Makefile.am b/tests/examples/Makefile.am
index aad6dc18..1199543e 100644
--- a/tests/examples/Makefile.am
+++ b/tests/examples/Makefile.am
@@ -1,3 +1,3 @@
-SUBDIRS = equalizer level rtp spectrum
+SUBDIRS = audiofx equalizer level rtp spectrum
 
-DIST_SUBDIRS = equalizer level rtp spectrum
+DIST_SUBDIRS = audiofx equalizer level rtp spectrum
diff --git a/tests/examples/audiofx/Makefile.am b/tests/examples/audiofx/Makefile.am
new file mode 100644
index 00000000..efcf785b
--- /dev/null
+++ b/tests/examples/audiofx/Makefile.am
@@ -0,0 +1,7 @@
+noinst_PROGRAMS = firfilter-example iirfilter-example
+
+firfilter_example_CFLAGS = $(GST_CFLAGS) $(GST_PLUGINS_BASE_CFLAGS)
+firfilter_example_LDADD = $(GST_LIBS) $(GST_PLUGINS_BASE_LIBS) -lgstfft-@GST_MAJORMINOR@ $(LIBM)
+
+iirfilter_example_CFLAGS = $(GST_CFLAGS)
+iirfilter_example_LDADD = $(GST_LIBS) $(LIBM)
diff --git a/tests/examples/audiofx/firfilter-example.c b/tests/examples/audiofx/firfilter-example.c
new file mode 100644
index 00000000..c5d56da1
--- /dev/null
+++ b/tests/examples/audiofx/firfilter-example.c
@@ -0,0 +1,161 @@
+/* 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 bandpass FIR filter
+ * by transforming the frequency response to the filter kernel.
+ */
+
+#include <string.h>
+#include <math.h>
+
+#include <gst/gst.h>
+#include <gst/fft/gstfftf64.h>
+
+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, };
+  GstFFTF64 *fft;
+  GstFFTF64Complex frequency_response[17];
+  gdouble tmp[32];
+  gdouble filter_kernel[32];
+  guint i;
+
+  /* Create the frequency response: zero outside
+   * a small frequency band */
+  for (i = 0; i < 17; i++) {
+    if (i < 5 || i > 11)
+      frequency_response[i].r = 0.0;
+    else
+      frequency_response[i].r = 1.0;
+
+    frequency_response[i].i = 0.0;
+  }
+
+  /* Calculate the inverse FT of the frequency response */
+  fft = gst_fft_f64_new (32, TRUE);
+  gst_fft_f64_inverse_fft (fft, frequency_response, tmp);
+  gst_fft_f64_free (fft);
+
+  /* Shift the inverse FT of the frequency response by 16,
+   * i.e. the half of the kernel length to get the
+   * impulse response. See http://www.dspguide.com/ch17/1.htm
+   * for more information.
+   */
+  for (i = 0; i < 32; i++)
+    filter_kernel[i] = tmp[(i + 16) % 32];
+
+  /* Apply the hamming window to the impulse response to get
+   * a better result than given from the rectangular window
+   */
+  for (i = 0; i < 32; i++)
+    filter_kernel[i] *= (0.54 - 0.46 * cos (2 * M_PI * i / 32));
+
+  va = g_value_array_new (1);
+
+  g_value_init (&v, G_TYPE_DOUBLE);
+  for (i = 0; i < 32; i++) {
+    g_value_set_double (&v, filter_kernel[i]);
+    g_value_array_append (va, &v);
+    g_value_reset (&v);
+  }
+  g_object_set (G_OBJECT (element), "kernel", va, NULL);
+  /* Latency is 1/2 of the kernel length for this method of
+   * calculating a filter kernel from the frequency response
+   */
+  g_object_set (G_OBJECT (element), "latency", 32 / 2, 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 ("audiofirfilter", 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;
+}
diff --git a/tests/examples/audiofx/iirfilter-example.c b/tests/examples/audiofx/iirfilter-example.c
new file mode 100644
index 00000000..8ccffb74
--- /dev/null
+++ b/tests/examples/audiofx/iirfilter-example.c
@@ -0,0 +1,137 @@
+/* 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;
+}