summaryrefslogtreecommitdiffstats
path: root/ext/gconf/gstgconfaudiosrc.c
blob: f5c349249767ea741e5705ebe131a109cde328b5 (plain)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
/* GStreamer
 * (c) 2005 Ronald S. Bultje <rbultje@ronald.bitfreak.net>
 * (c) 2005 Tim-Philipp Müller <tim centricular net>
 *
 * 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.
 */
/**
 * SECTION:element-gconfaudiosrc
 * @see_also: #GstAlsaSrc, #GstAutoAudioSrc
 *
 * This element records sound from the audiosink that has been configured in
 * GConf by the user.
 *
 * <refsect2>
 * <title>Example launch line</title>
 * |[
 * gst-launch gconfaudiosrc ! audioconvert ! wavenc ! filesink location=record.wav
 * ]| Record from configured audioinput
 * </refsect2>
 */

#ifdef HAVE_CONFIG_H
#include "config.h"
#endif

#include <string.h>

#include "gstgconfelements.h"
#include "gstgconfaudiosrc.h"

static void gst_gconf_audio_src_dispose (GObject * object);
static void gst_gconf_audio_src_finalize (GstGConfAudioSrc * src);
static void cb_toggle_element (GConfClient * client,
    guint connection_id, GConfEntry * entry, gpointer data);
static GstStateChangeReturn
gst_gconf_audio_src_change_state (GstElement * element,
    GstStateChange transition);

GST_BOILERPLATE (GstGConfAudioSrc, gst_gconf_audio_src, GstBin, GST_TYPE_BIN);

static void
gst_gconf_audio_src_base_init (gpointer klass)
{
  GstElementClass *eklass = GST_ELEMENT_CLASS (klass);
  static const GstElementDetails gst_gconf_audio_src_details =
      GST_ELEMENT_DETAILS ("GConf audio source",
      "Source/Audio",
      "Audio source embedding the GConf-settings for audio input",
      "GStreamer maintainers <gstreamer-devel@lists.sourceforge.net>");
  static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
      GST_PAD_SRC,
      GST_PAD_ALWAYS,
      GST_STATIC_CAPS_ANY);

  gst_element_class_add_pad_template (eklass,
      gst_static_pad_template_get (&src_template));
  gst_element_class_set_details (eklass, &gst_gconf_audio_src_details);
}

static void
gst_gconf_audio_src_class_init (GstGConfAudioSrcClass * klass)
{
  GObjectClass *oklass = G_OBJECT_CLASS (klass);
  GstElementClass *eklass = GST_ELEMENT_CLASS (klass);

  oklass->dispose = gst_gconf_audio_src_dispose;
  oklass->finalize = (GObjectFinalizeFunc) gst_gconf_audio_src_finalize;
  eklass->change_state = gst_gconf_audio_src_change_state;
}

/*
 * Hack to make negotiation work.
 */

static gboolean
gst_gconf_audio_src_reset (GstGConfAudioSrc * src)
{
  GstPad *targetpad;

  /* fakesrc */
  if (src->kid) {
    gst_element_set_state (src->kid, GST_STATE_NULL);
    gst_bin_remove (GST_BIN (src), src->kid);
  }
  src->kid = gst_element_factory_make ("fakesrc", "testsrc");
  if (!src->kid) {
    GST_ERROR_OBJECT (src, "Failed to create fakesrc");
    return FALSE;
  }
  gst_bin_add (GST_BIN (src), src->kid);

  targetpad = gst_element_get_static_pad (src->kid, "src");
  gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
  gst_object_unref (targetpad);

  g_free (src->gconf_str);
  src->gconf_str = NULL;
  return TRUE;
}

static void
gst_gconf_audio_src_init (GstGConfAudioSrc * src,
    GstGConfAudioSrcClass * g_class)
{
  src->pad = gst_ghost_pad_new_no_target ("src", GST_PAD_SRC);
  gst_element_add_pad (GST_ELEMENT (src), src->pad);

  gst_gconf_audio_src_reset (src);

  src->client = gconf_client_get_default ();
  gconf_client_add_dir (src->client, GST_GCONF_DIR,
      GCONF_CLIENT_PRELOAD_RECURSIVE, NULL);
  src->gconf_notify_id = gconf_client_notify_add (src->client,
      GST_GCONF_DIR "/" GST_GCONF_AUDIOSRC_KEY,
      cb_toggle_element, src, NULL, NULL);
}

static void
gst_gconf_audio_src_dispose (GObject * object)
{
  GstGConfAudioSrc *src = GST_GCONF_AUDIO_SRC (object);

  if (src->client) {
    if (src->gconf_notify_id) {
      gconf_client_notify_remove (src->client, src->gconf_notify_id);
      src->gconf_notify_id = 0;
    }

    g_object_unref (G_OBJECT (src->client));
    src->client = NULL;
  }

  GST_CALL_PARENT (G_OBJECT_CLASS, dispose, (object));
}

static void
gst_gconf_audio_src_finalize (GstGConfAudioSrc * src)
{
  g_free (src->gconf_str);

  GST_CALL_PARENT (G_OBJECT_CLASS, finalize, ((GObject *) (src)));
}

static gboolean
do_toggle_element (GstGConfAudioSrc * src)
{
  GstState cur, next;
  GstPad *targetpad;
  gchar *new_gconf_str;

  new_gconf_str = gst_gconf_get_string (GST_GCONF_AUDIOSRC_KEY);
  if (new_gconf_str != NULL && src->gconf_str != NULL &&
      (strlen (new_gconf_str) == 0 ||
          strcmp (src->gconf_str, new_gconf_str) == 0)) {
    g_free (new_gconf_str);
    GST_DEBUG_OBJECT (src, "GConf key was updated, but it didn't change");
    return TRUE;
  }

  GST_OBJECT_LOCK (src);
  cur = GST_STATE (src);
  next = GST_STATE_PENDING (src);
  GST_OBJECT_UNLOCK (src);

  if (cur >= GST_STATE_READY || next == GST_STATE_PAUSED) {
    GST_DEBUG_OBJECT (src, "already running, ignoring GConf change");
    g_free (new_gconf_str);
    return TRUE;
  }

  GST_DEBUG_OBJECT (src, "GConf key changed: '%s' to '%s'",
      GST_STR_NULL (src->gconf_str), GST_STR_NULL (new_gconf_str));

  g_free (src->gconf_str);
  src->gconf_str = new_gconf_str;

  /* kill old element */
  if (src->kid) {
    GST_DEBUG_OBJECT (src, "Removing old kid");
    gst_element_set_state (src->kid, GST_STATE_NULL);
    gst_bin_remove (GST_BIN (src), src->kid);
    src->kid = NULL;
  }

  GST_DEBUG_OBJECT (src, "Creating new kid");
  if (!(src->kid = gst_gconf_get_default_audio_src ())) {
    GST_ELEMENT_ERROR (src, LIBRARY, SETTINGS, (NULL),
        ("Failed to render audio source from GConf"));
    g_free (src->gconf_str);
    src->gconf_str = NULL;
    return FALSE;
  }
  gst_element_set_state (src->kid, GST_STATE (src));
  gst_bin_add (GST_BIN (src), src->kid);

  /* re-attach ghostpad */
  GST_DEBUG_OBJECT (src, "Creating new ghostpad");
  targetpad = gst_element_get_static_pad (src->kid, "src");
  gst_ghost_pad_set_target (GST_GHOST_PAD (src->pad), targetpad);
  gst_object_unref (targetpad);
  GST_DEBUG_OBJECT (src, "done changing gconf audio source");

  return TRUE;
}

static void
cb_toggle_element (GConfClient * client,
    guint connection_id, GConfEntry * entry, gpointer data)
{
  do_toggle_element (GST_GCONF_AUDIO_SRC (data));
}

static GstStateChangeReturn
gst_gconf_audio_src_change_state (GstElement * element,
    GstStateChange transition)
{
  GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
  GstGConfAudioSrc *src = GST_GCONF_AUDIO_SRC (element);

  switch (transition) {
    case GST_STATE_CHANGE_NULL_TO_READY:
      if (!do_toggle_element (src))
        return GST_STATE_CHANGE_FAILURE;
      break;
    default:
      break;
  }

  ret = GST_CALL_PARENT_WITH_DEFAULT (GST_ELEMENT_CLASS, change_state,
      (element, transition), GST_STATE_CHANGE_SUCCESS);

  switch (transition) {
    case GST_STATE_CHANGE_READY_TO_NULL:
      if (!gst_gconf_audio_src_reset (src))
        ret = GST_STATE_CHANGE_FAILURE;
      break;
    default:
      break;
  }

  return ret;
}