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1 /* GStreamer |
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2 * Copyright (C) 1999,2000 Erik Walthinsen <omega@cse.ogi.edu> |
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3 * 2005 Wim Taymans <wim@fluendo.com> |
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4 * |
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5 * gstaudiosink.c: simple audio sink base class |
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6 * |
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7 * This library is free software; you can redistribute it and/or |
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8 * modify it under the terms of the GNU Library General Public |
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9 * License as published by the Free Software Foundation; either |
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10 * version 2 of the License, or (at your option) any later version. |
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11 * |
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12 * This library is distributed in the hope that it will be useful, |
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13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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15 * Library General Public License for more details. |
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16 * |
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17 * You should have received a copy of the GNU Library General Public |
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18 * License along with this library; if not, write to the |
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19 * Free Software Foundation, Inc., 59 Temple Place - Suite 330, |
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20 * Boston, MA 02111-1307, USA. |
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21 */ |
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22 |
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23 /** |
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24 * SECTION:gstaudiosink |
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25 * @short_description: Simple base class for audio sinks |
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26 * @see_also: #GstBaseAudioSink, #GstRingBuffer, #GstAudioSink. |
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27 * |
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28 * This is the most simple base class for audio sinks that only requires |
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29 * subclasses to implement a set of simple functions: |
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30 * |
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31 * <variablelist> |
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32 * <varlistentry> |
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33 * <term>open()</term> |
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34 * <listitem><para>Open the device.</para></listitem> |
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35 * </varlistentry> |
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36 * <varlistentry> |
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37 * <term>prepare()</term> |
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38 * <listitem><para>Configure the device with the specified format.</para></listitem> |
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39 * </varlistentry> |
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40 * <varlistentry> |
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41 * <term>write()</term> |
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42 * <listitem><para>Write samples to the device.</para></listitem> |
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43 * </varlistentry> |
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44 * <varlistentry> |
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45 * <term>reset()</term> |
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46 * <listitem><para>Unblock writes and flush the device.</para></listitem> |
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47 * </varlistentry> |
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48 * <varlistentry> |
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49 * <term>delay()</term> |
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50 * <listitem><para>Get the number of samples written but not yet played |
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51 * by the device.</para></listitem> |
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52 * </varlistentry> |
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53 * <varlistentry> |
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54 * <term>unprepare()</term> |
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55 * <listitem><para>Undo operations done by prepare.</para></listitem> |
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56 * </varlistentry> |
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57 * <varlistentry> |
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58 * <term>close()</term> |
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59 * <listitem><para>Close the device.</para></listitem> |
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60 * </varlistentry> |
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61 * </variablelist> |
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62 * |
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63 * All scheduling of samples and timestamps is done in this base class |
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64 * together with #GstBaseAudioSink using a default implementation of a |
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65 * #GstRingBuffer that uses threads. |
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66 * |
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67 * Last reviewed on 2006-09-27 (0.10.12) |
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68 */ |
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69 |
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70 #include <string.h> |
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71 |
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72 #include "gstaudiosink.h" |
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73 |
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74 #ifdef __SYMBIAN32__ |
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75 #include <glib_global.h> |
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76 #endif |
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77 |
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78 GST_DEBUG_CATEGORY_STATIC (gst_audio_sink_debug); |
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79 #define GST_CAT_DEFAULT gst_audio_sink_debug |
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80 |
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81 #define GST_TYPE_AUDIORING_BUFFER \ |
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82 (gst_audioringbuffer_get_type()) |
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83 #define GST_AUDIORING_BUFFER(obj) \ |
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84 (G_TYPE_CHECK_INSTANCE_CAST((obj),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBuffer)) |
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85 #define GST_AUDIORING_BUFFER_CLASS(klass) \ |
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86 (G_TYPE_CHECK_CLASS_CAST((klass),GST_TYPE_AUDIORING_BUFFER,GstAudioRingBufferClass)) |
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87 #define GST_AUDIORING_BUFFER_GET_CLASS(obj) \ |
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88 (G_TYPE_INSTANCE_GET_CLASS ((obj), GST_TYPE_AUDIORING_BUFFER, GstAudioRingBufferClass)) |
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89 #define GST_AUDIORING_BUFFER_CAST(obj) \ |
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90 ((GstAudioRingBuffer *)obj) |
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91 #define GST_IS_AUDIORING_BUFFER(obj) \ |
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92 (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORING_BUFFER)) |
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93 #define GST_IS_AUDIORING_BUFFER_CLASS(klass)\ |
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94 (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORING_BUFFER)) |
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95 |
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96 typedef struct _GstAudioRingBuffer GstAudioRingBuffer; |
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97 typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass; |
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98 |
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99 #define GST_AUDIORING_BUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond) |
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100 #define GST_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf))) |
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101 #define GST_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORING_BUFFER_GET_COND (buf))) |
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102 #define GST_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORING_BUFFER_GET_COND (buf))) |
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103 |
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104 struct _GstAudioRingBuffer |
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105 { |
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106 GstRingBuffer object; |
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107 |
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108 gboolean running; |
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109 gint queuedseg; |
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110 |
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111 GCond *cond; |
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112 }; |
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113 |
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114 struct _GstAudioRingBufferClass |
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115 { |
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116 GstRingBufferClass parent_class; |
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117 }; |
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118 |
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119 static void gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass); |
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120 static void gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer, |
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121 GstAudioRingBufferClass * klass); |
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122 static void gst_audioringbuffer_dispose (GObject * object); |
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123 static void gst_audioringbuffer_finalize (GObject * object); |
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124 |
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125 static GstRingBufferClass *ring_parent_class = NULL; |
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126 |
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127 static gboolean gst_audioringbuffer_open_device (GstRingBuffer * buf); |
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128 static gboolean gst_audioringbuffer_close_device (GstRingBuffer * buf); |
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129 static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf, |
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130 GstRingBufferSpec * spec); |
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131 static gboolean gst_audioringbuffer_release (GstRingBuffer * buf); |
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132 static gboolean gst_audioringbuffer_start (GstRingBuffer * buf); |
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133 static gboolean gst_audioringbuffer_pause (GstRingBuffer * buf); |
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134 static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf); |
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135 static guint gst_audioringbuffer_delay (GstRingBuffer * buf); |
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136 |
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137 /* ringbuffer abstract base class */ |
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138 static GType |
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139 gst_audioringbuffer_get_type (void) |
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140 { |
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141 static GType ringbuffer_type = 0; |
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142 |
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143 if (!ringbuffer_type) { |
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144 static const GTypeInfo ringbuffer_info = { |
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145 sizeof (GstAudioRingBufferClass), |
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146 NULL, |
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147 NULL, |
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148 (GClassInitFunc) gst_audioringbuffer_class_init, |
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149 NULL, |
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150 NULL, |
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151 sizeof (GstAudioRingBuffer), |
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152 0, |
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153 (GInstanceInitFunc) gst_audioringbuffer_init, |
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154 NULL |
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155 }; |
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156 |
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157 ringbuffer_type = |
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158 g_type_register_static (GST_TYPE_RING_BUFFER, "GstAudioSinkRingBuffer", |
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159 &ringbuffer_info, 0); |
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160 } |
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161 return ringbuffer_type; |
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162 } |
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163 |
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164 static void |
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165 gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass) |
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166 { |
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167 GObjectClass *gobject_class; |
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168 GstObjectClass *gstobject_class; |
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169 GstRingBufferClass *gstringbuffer_class; |
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170 |
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171 gobject_class = (GObjectClass *) klass; |
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172 gstobject_class = (GstObjectClass *) klass; |
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173 gstringbuffer_class = (GstRingBufferClass *) klass; |
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174 |
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175 ring_parent_class = g_type_class_peek_parent (klass); |
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176 |
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177 gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_audioringbuffer_dispose); |
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178 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audioringbuffer_finalize); |
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179 |
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180 gstringbuffer_class->open_device = |
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181 GST_DEBUG_FUNCPTR (gst_audioringbuffer_open_device); |
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182 gstringbuffer_class->close_device = |
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183 GST_DEBUG_FUNCPTR (gst_audioringbuffer_close_device); |
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184 gstringbuffer_class->acquire = |
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185 GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire); |
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186 gstringbuffer_class->release = |
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187 GST_DEBUG_FUNCPTR (gst_audioringbuffer_release); |
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188 gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start); |
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189 gstringbuffer_class->pause = GST_DEBUG_FUNCPTR (gst_audioringbuffer_pause); |
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190 gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start); |
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191 gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop); |
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192 |
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193 gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay); |
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194 } |
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195 |
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196 typedef guint (*WriteFunc) (GstAudioSink * sink, gpointer data, guint length); |
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197 |
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198 /* this internal thread does nothing else but write samples to the audio device. |
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199 * It will write each segment in the ringbuffer and will update the play |
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200 * pointer. |
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201 * The start/stop methods control the thread. |
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202 */ |
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203 static void |
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204 audioringbuffer_thread_func (GstRingBuffer * buf) |
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205 { |
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206 GstAudioSink *sink; |
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207 GstAudioSinkClass *csink; |
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208 GstAudioRingBuffer *abuf = GST_AUDIORING_BUFFER_CAST (buf); |
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209 WriteFunc writefunc; |
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210 |
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211 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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212 csink = GST_AUDIO_SINK_GET_CLASS (sink); |
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213 |
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214 GST_DEBUG_OBJECT (sink, "enter thread"); |
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215 |
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216 writefunc = csink->write; |
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217 if (writefunc == NULL) |
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218 goto no_function; |
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219 |
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220 while (TRUE) { |
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221 gint left, len; |
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222 guint8 *readptr; |
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223 gint readseg; |
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224 |
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225 if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) { |
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226 gint written = 0; |
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227 |
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228 left = len; |
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229 do { |
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230 written = writefunc (sink, readptr + written, left); |
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231 GST_LOG_OBJECT (sink, "transfered %d bytes of %d from segment %d", |
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232 written, left, readseg); |
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233 if (written < 0 || written > left) { |
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234 GST_WARNING_OBJECT (sink, |
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235 "error writing data (reason: %s), skipping segment", |
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236 g_strerror (errno)); |
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237 break; |
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238 } |
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239 left -= written; |
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240 } while (left > 0); |
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241 |
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242 /* clear written samples */ |
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243 gst_ring_buffer_clear (buf, readseg); |
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244 |
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245 /* we wrote one segment */ |
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246 gst_ring_buffer_advance (buf, 1); |
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247 } else { |
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248 GST_OBJECT_LOCK (abuf); |
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249 if (!abuf->running) |
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250 goto stop_running; |
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251 GST_DEBUG_OBJECT (sink, "signal wait"); |
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252 GST_AUDIORING_BUFFER_SIGNAL (buf); |
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253 GST_DEBUG_OBJECT (sink, "wait for action"); |
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254 GST_AUDIORING_BUFFER_WAIT (buf); |
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255 GST_DEBUG_OBJECT (sink, "got signal"); |
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256 if (!abuf->running) |
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257 goto stop_running; |
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258 GST_DEBUG_OBJECT (sink, "continue running"); |
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259 GST_OBJECT_UNLOCK (abuf); |
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260 } |
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261 } |
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262 GST_DEBUG_OBJECT (sink, "exit thread"); |
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263 |
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264 return; |
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265 |
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266 /* ERROR */ |
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267 no_function: |
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268 { |
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269 GST_DEBUG_OBJECT (sink, "no write function, exit thread"); |
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270 return; |
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271 } |
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272 stop_running: |
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273 { |
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274 GST_OBJECT_UNLOCK (abuf); |
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275 GST_DEBUG_OBJECT (sink, "stop running, exit thread"); |
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276 return; |
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277 } |
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278 } |
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279 |
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280 static void |
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281 gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer, |
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282 GstAudioRingBufferClass * g_class) |
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283 { |
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284 ringbuffer->running = FALSE; |
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285 ringbuffer->queuedseg = 0; |
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286 |
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287 ringbuffer->cond = g_cond_new (); |
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288 } |
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289 |
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290 static void |
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291 gst_audioringbuffer_dispose (GObject * object) |
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292 { |
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293 G_OBJECT_CLASS (ring_parent_class)->dispose (object); |
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294 } |
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295 |
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296 static void |
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297 gst_audioringbuffer_finalize (GObject * object) |
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298 { |
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299 GstAudioRingBuffer *ringbuffer = GST_AUDIORING_BUFFER_CAST (object); |
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300 |
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301 g_cond_free (ringbuffer->cond); |
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302 |
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303 G_OBJECT_CLASS (ring_parent_class)->finalize (object); |
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304 } |
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305 |
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306 static gboolean |
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307 gst_audioringbuffer_open_device (GstRingBuffer * buf) |
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308 { |
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309 GstAudioSink *sink; |
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310 GstAudioSinkClass *csink; |
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311 gboolean result = TRUE; |
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312 |
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313 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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314 csink = GST_AUDIO_SINK_GET_CLASS (sink); |
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315 |
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316 if (csink->open) |
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317 result = csink->open (sink); |
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318 |
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319 if (!result) |
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320 goto could_not_open; |
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321 |
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322 return result; |
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323 |
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324 could_not_open: |
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325 { |
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326 GST_DEBUG_OBJECT (sink, "could not open device"); |
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327 return FALSE; |
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328 } |
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329 } |
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330 |
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331 static gboolean |
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332 gst_audioringbuffer_close_device (GstRingBuffer * buf) |
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333 { |
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334 GstAudioSink *sink; |
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335 GstAudioSinkClass *csink; |
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336 gboolean result = TRUE; |
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337 |
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338 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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339 csink = GST_AUDIO_SINK_GET_CLASS (sink); |
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340 |
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341 if (csink->close) |
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342 result = csink->close (sink); |
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343 |
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344 if (!result) |
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345 goto could_not_close; |
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346 |
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347 return result; |
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348 |
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349 could_not_close: |
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350 { |
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351 GST_DEBUG_OBJECT (sink, "could not close device"); |
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352 return FALSE; |
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353 } |
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354 } |
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355 |
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356 static gboolean |
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357 gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) |
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358 { |
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359 GstAudioSink *sink; |
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360 GstAudioSinkClass *csink; |
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361 GstAudioRingBuffer *abuf; |
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362 gboolean result = FALSE; |
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363 |
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364 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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365 csink = GST_AUDIO_SINK_GET_CLASS (sink); |
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366 |
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367 if (csink->prepare) |
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368 result = csink->prepare (sink, spec); |
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369 |
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370 if (!result) |
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371 goto could_not_prepare; |
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372 |
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373 /* allocate one more segment as we need some headroom */ |
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374 spec->segtotal++; |
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375 |
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376 buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); |
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377 memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); |
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378 |
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379 abuf = GST_AUDIORING_BUFFER_CAST (buf); |
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380 abuf->running = TRUE; |
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381 |
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382 sink->thread = |
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383 g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE, |
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384 NULL); |
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385 GST_AUDIORING_BUFFER_WAIT (buf); |
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386 |
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387 return result; |
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388 |
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389 could_not_prepare: |
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390 { |
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391 GST_DEBUG_OBJECT (sink, "could not prepare device"); |
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392 return FALSE; |
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393 } |
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394 } |
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395 |
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396 /* function is called with LOCK */ |
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397 static gboolean |
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398 gst_audioringbuffer_release (GstRingBuffer * buf) |
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399 { |
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400 GstAudioSink *sink; |
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401 GstAudioSinkClass *csink; |
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402 GstAudioRingBuffer *abuf; |
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403 gboolean result = FALSE; |
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404 |
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405 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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406 csink = GST_AUDIO_SINK_GET_CLASS (sink); |
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407 abuf = GST_AUDIORING_BUFFER_CAST (buf); |
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408 |
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409 abuf->running = FALSE; |
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410 GST_DEBUG_OBJECT (sink, "signal wait"); |
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411 GST_AUDIORING_BUFFER_SIGNAL (buf); |
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412 GST_OBJECT_UNLOCK (buf); |
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413 |
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414 /* join the thread */ |
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415 g_thread_join (sink->thread); |
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416 |
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417 GST_OBJECT_LOCK (buf); |
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418 |
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419 /* free the buffer */ |
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420 gst_buffer_unref (buf->data); |
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421 buf->data = NULL; |
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422 |
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423 if (csink->unprepare) |
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424 result = csink->unprepare (sink); |
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425 |
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426 if (!result) |
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427 goto could_not_unprepare; |
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428 |
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429 return result; |
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430 |
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431 could_not_unprepare: |
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432 { |
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433 GST_DEBUG_OBJECT (sink, "could not unprepare device"); |
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434 return FALSE; |
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435 } |
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436 } |
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437 |
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438 static gboolean |
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439 gst_audioringbuffer_start (GstRingBuffer * buf) |
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440 { |
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441 GstAudioSink *sink; |
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442 |
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443 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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444 |
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445 GST_DEBUG_OBJECT (sink, "start, sending signal"); |
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446 GST_AUDIORING_BUFFER_SIGNAL (buf); |
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447 |
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448 return TRUE; |
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449 } |
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450 |
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451 static gboolean |
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452 gst_audioringbuffer_pause (GstRingBuffer * buf) |
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453 { |
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454 GstAudioSink *sink; |
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455 GstAudioSinkClass *csink; |
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456 |
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457 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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458 csink = GST_AUDIO_SINK_GET_CLASS (sink); |
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459 |
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460 /* unblock any pending writes to the audio device */ |
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461 if (csink->reset) { |
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462 GST_DEBUG_OBJECT (sink, "reset..."); |
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463 csink->reset (sink); |
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464 GST_DEBUG_OBJECT (sink, "reset done"); |
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465 } |
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466 |
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467 return TRUE; |
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468 } |
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469 |
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470 static gboolean |
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471 gst_audioringbuffer_stop (GstRingBuffer * buf) |
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472 { |
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473 GstAudioSink *sink; |
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474 GstAudioSinkClass *csink; |
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475 GstAudioRingBuffer *abuf; |
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476 |
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477 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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478 csink = GST_AUDIO_SINK_GET_CLASS (sink); |
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479 abuf = GST_AUDIORING_BUFFER_CAST (buf); |
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480 |
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481 /* unblock any pending writes to the audio device */ |
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482 if (csink->reset) { |
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483 GST_DEBUG_OBJECT (sink, "reset..."); |
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484 csink->reset (sink); |
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485 GST_DEBUG_OBJECT (sink, "reset done"); |
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486 } |
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487 |
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488 if (abuf->running) { |
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489 GST_DEBUG_OBJECT (sink, "stop, waiting..."); |
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490 GST_AUDIORING_BUFFER_WAIT (buf); |
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491 GST_DEBUG_OBJECT (sink, "stopped"); |
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492 } |
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493 |
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494 return TRUE; |
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495 } |
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496 |
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497 static guint |
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498 gst_audioringbuffer_delay (GstRingBuffer * buf) |
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499 { |
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500 GstAudioSink *sink; |
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501 GstAudioSinkClass *csink; |
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502 guint res = 0; |
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503 |
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504 sink = GST_AUDIO_SINK (GST_OBJECT_PARENT (buf)); |
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505 csink = GST_AUDIO_SINK_GET_CLASS (sink); |
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506 |
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507 if (csink->delay) |
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508 res = csink->delay (sink); |
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509 |
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510 return res; |
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511 } |
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512 |
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513 /* AudioSink signals and args */ |
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514 enum |
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515 { |
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516 /* FILL ME */ |
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517 LAST_SIGNAL |
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518 }; |
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519 |
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520 enum |
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521 { |
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522 ARG_0, |
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523 }; |
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524 |
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525 #define _do_init(bla) \ |
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526 GST_DEBUG_CATEGORY_INIT (gst_audio_sink_debug, "audiosink", 0, "audiosink element"); |
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527 |
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528 GST_BOILERPLATE_FULL (GstAudioSink, gst_audio_sink, GstBaseAudioSink, |
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529 GST_TYPE_BASE_AUDIO_SINK, _do_init); |
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530 |
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531 static GstRingBuffer *gst_audio_sink_create_ringbuffer (GstBaseAudioSink * |
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532 sink); |
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533 |
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534 static void |
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535 gst_audio_sink_base_init (gpointer g_class) |
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536 { |
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537 } |
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538 |
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539 static void |
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540 gst_audio_sink_class_init (GstAudioSinkClass * klass) |
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541 { |
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542 GObjectClass *gobject_class; |
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543 GstElementClass *gstelement_class; |
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544 GstBaseSinkClass *gstbasesink_class; |
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545 GstBaseAudioSinkClass *gstbaseaudiosink_class; |
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546 |
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547 gobject_class = (GObjectClass *) klass; |
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548 gstelement_class = (GstElementClass *) klass; |
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549 gstbasesink_class = (GstBaseSinkClass *) klass; |
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550 gstbaseaudiosink_class = (GstBaseAudioSinkClass *) klass; |
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551 |
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552 gstbaseaudiosink_class->create_ringbuffer = |
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553 GST_DEBUG_FUNCPTR (gst_audio_sink_create_ringbuffer); |
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554 } |
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555 |
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556 static void |
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557 gst_audio_sink_init (GstAudioSink * audiosink, GstAudioSinkClass * g_class) |
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558 { |
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559 } |
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560 |
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561 static GstRingBuffer * |
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562 gst_audio_sink_create_ringbuffer (GstBaseAudioSink * sink) |
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563 { |
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564 GstRingBuffer *buffer; |
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565 |
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566 GST_DEBUG_OBJECT (sink, "creating ringbuffer"); |
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567 buffer = g_object_new (GST_TYPE_AUDIORING_BUFFER, NULL); |
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568 GST_DEBUG_OBJECT (sink, "created ringbuffer @%p", buffer); |
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569 |
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570 return buffer; |
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571 } |