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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 * gstaudiosrc.c: simple audio src 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:gstaudiosrc |
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25 * @short_description: Simple base class for audio sources |
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26 * @see_also: #GstBaseAudioSrc, #GstRingBuffer, #GstAudioSrc. |
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27 * |
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28 * This is the most simple base class for audio sources 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>read()</term> |
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42 * <listitem><para>Read samples from 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 reads 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 in the device but not yet read. |
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51 * </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 #GstBaseAudioSrc 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 "gstaudiosrc.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_src_debug); |
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79 #define GST_CAT_DEFAULT gst_audio_src_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_IS_AUDIORING_BUFFER(obj) \ |
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90 (G_TYPE_CHECK_INSTANCE_TYPE((obj),GST_TYPE_AUDIORING_BUFFER)) |
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91 #define GST_IS_AUDIORING_BUFFER_CLASS(klass)\ |
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92 (G_TYPE_CHECK_CLASS_TYPE((klass),GST_TYPE_AUDIORING_BUFFER)) |
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93 |
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94 typedef struct _GstAudioRingBuffer GstAudioRingBuffer; |
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95 typedef struct _GstAudioRingBufferClass GstAudioRingBufferClass; |
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96 |
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97 #define GST_AUDIORING_BUFFER_GET_COND(buf) (((GstAudioRingBuffer *)buf)->cond) |
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98 #define GST_AUDIORING_BUFFER_WAIT(buf) (g_cond_wait (GST_AUDIORING_BUFFER_GET_COND (buf), GST_OBJECT_GET_LOCK (buf))) |
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99 #define GST_AUDIORING_BUFFER_SIGNAL(buf) (g_cond_signal (GST_AUDIORING_BUFFER_GET_COND (buf))) |
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100 #define GST_AUDIORING_BUFFER_BROADCAST(buf)(g_cond_broadcast (GST_AUDIORING_BUFFER_GET_COND (buf))) |
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101 |
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102 struct _GstAudioRingBuffer |
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103 { |
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104 GstRingBuffer object; |
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105 |
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106 gboolean running; |
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107 gint queuedseg; |
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108 |
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109 GCond *cond; |
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110 }; |
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111 |
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112 struct _GstAudioRingBufferClass |
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113 { |
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114 GstRingBufferClass parent_class; |
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115 }; |
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116 |
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117 static void gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass); |
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118 static void gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer, |
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119 GstAudioRingBufferClass * klass); |
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120 static void gst_audioringbuffer_dispose (GObject * object); |
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121 static void gst_audioringbuffer_finalize (GObject * object); |
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122 |
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123 static GstRingBufferClass *ring_parent_class = NULL; |
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124 |
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125 static gboolean gst_audioringbuffer_open_device (GstRingBuffer * buf); |
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126 static gboolean gst_audioringbuffer_close_device (GstRingBuffer * buf); |
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127 static gboolean gst_audioringbuffer_acquire (GstRingBuffer * buf, |
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128 GstRingBufferSpec * spec); |
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129 static gboolean gst_audioringbuffer_release (GstRingBuffer * buf); |
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130 static gboolean gst_audioringbuffer_start (GstRingBuffer * buf); |
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131 static gboolean gst_audioringbuffer_stop (GstRingBuffer * buf); |
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132 static guint gst_audioringbuffer_delay (GstRingBuffer * buf); |
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133 |
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134 /* ringbuffer abstract base class */ |
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135 static GType |
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136 gst_audioringbuffer_get_type (void) |
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137 { |
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138 static GType ringbuffer_type = 0; |
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139 |
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140 if (!ringbuffer_type) { |
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141 static const GTypeInfo ringbuffer_info = { |
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142 sizeof (GstAudioRingBufferClass), |
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143 NULL, |
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144 NULL, |
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145 (GClassInitFunc) gst_audioringbuffer_class_init, |
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146 NULL, |
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147 NULL, |
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148 sizeof (GstAudioRingBuffer), |
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149 0, |
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150 (GInstanceInitFunc) gst_audioringbuffer_init, |
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151 NULL |
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152 }; |
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153 |
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154 ringbuffer_type = |
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155 g_type_register_static (GST_TYPE_RING_BUFFER, "GstAudioSrcRingBuffer", |
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156 &ringbuffer_info, 0); |
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157 } |
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158 return ringbuffer_type; |
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159 } |
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160 |
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161 static void |
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162 gst_audioringbuffer_class_init (GstAudioRingBufferClass * klass) |
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163 { |
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164 GObjectClass *gobject_class; |
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165 GstObjectClass *gstobject_class; |
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166 GstRingBufferClass *gstringbuffer_class; |
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167 |
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168 gobject_class = (GObjectClass *) klass; |
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169 gstobject_class = (GstObjectClass *) klass; |
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170 gstringbuffer_class = (GstRingBufferClass *) klass; |
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171 |
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172 ring_parent_class = g_type_class_peek_parent (klass); |
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173 |
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174 gobject_class->dispose = GST_DEBUG_FUNCPTR (gst_audioringbuffer_dispose); |
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175 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_audioringbuffer_finalize); |
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176 |
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177 gstringbuffer_class->open_device = |
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178 GST_DEBUG_FUNCPTR (gst_audioringbuffer_open_device); |
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179 gstringbuffer_class->close_device = |
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180 GST_DEBUG_FUNCPTR (gst_audioringbuffer_close_device); |
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181 gstringbuffer_class->acquire = |
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182 GST_DEBUG_FUNCPTR (gst_audioringbuffer_acquire); |
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183 gstringbuffer_class->release = |
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184 GST_DEBUG_FUNCPTR (gst_audioringbuffer_release); |
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185 gstringbuffer_class->start = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start); |
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186 gstringbuffer_class->resume = GST_DEBUG_FUNCPTR (gst_audioringbuffer_start); |
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187 gstringbuffer_class->stop = GST_DEBUG_FUNCPTR (gst_audioringbuffer_stop); |
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188 |
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189 gstringbuffer_class->delay = GST_DEBUG_FUNCPTR (gst_audioringbuffer_delay); |
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190 } |
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191 |
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192 typedef guint (*ReadFunc) (GstAudioSrc * src, gpointer data, guint length); |
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193 |
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194 /* this internal thread does nothing else but read samples from the audio device. |
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195 * It will read each segment in the ringbuffer and will update the play |
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196 * pointer. |
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197 * The start/stop methods control the thread. |
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198 */ |
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199 static void |
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200 audioringbuffer_thread_func (GstRingBuffer * buf) |
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201 { |
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202 GstAudioSrc *src; |
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203 GstAudioSrcClass *csrc; |
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204 GstAudioRingBuffer *abuf = GST_AUDIORING_BUFFER (buf); |
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205 ReadFunc readfunc; |
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206 |
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207 src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf)); |
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208 csrc = GST_AUDIO_SRC_GET_CLASS (src); |
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209 |
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210 GST_DEBUG_OBJECT (src, "enter thread"); |
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211 |
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212 readfunc = csrc->read; |
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213 if (readfunc == NULL) |
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214 goto no_function; |
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215 |
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216 while (TRUE) { |
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217 gint left, len; |
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218 guint8 *readptr; |
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219 gint readseg; |
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220 |
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221 if (gst_ring_buffer_prepare_read (buf, &readseg, &readptr, &len)) { |
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222 gint read = 0; |
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223 |
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224 left = len; |
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225 do { |
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226 read = readfunc (src, readptr + read, left); |
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227 GST_LOG_OBJECT (src, "transfered %d bytes of %d to segment %d", read, |
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228 left, readseg); |
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229 if (read < 0 || read > left) { |
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230 GST_WARNING_OBJECT (src, |
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231 "error reading data (reason: %s), skipping segment", |
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232 g_strerror (errno)); |
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233 break; |
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234 } |
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235 left -= read; |
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236 } while (left > 0); |
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237 |
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238 /* we read one segment */ |
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239 gst_ring_buffer_advance (buf, 1); |
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240 } else { |
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241 GST_OBJECT_LOCK (abuf); |
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242 if (!abuf->running) |
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243 goto stop_running; |
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244 GST_DEBUG_OBJECT (src, "signal wait"); |
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245 GST_AUDIORING_BUFFER_SIGNAL (buf); |
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246 GST_DEBUG_OBJECT (src, "wait for action"); |
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247 GST_AUDIORING_BUFFER_WAIT (buf); |
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248 GST_DEBUG_OBJECT (src, "got signal"); |
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249 if (!abuf->running) |
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250 goto stop_running; |
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251 GST_DEBUG_OBJECT (src, "continue running"); |
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252 GST_OBJECT_UNLOCK (abuf); |
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253 } |
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254 } |
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255 GST_DEBUG_OBJECT (src, "exit thread"); |
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256 |
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257 return; |
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258 |
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259 /* ERROR */ |
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260 no_function: |
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261 { |
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262 GST_DEBUG ("no write function, exit thread"); |
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263 return; |
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264 } |
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265 stop_running: |
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266 { |
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267 GST_OBJECT_UNLOCK (abuf); |
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268 GST_DEBUG ("stop running, exit thread"); |
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269 return; |
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270 } |
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271 } |
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272 |
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273 static void |
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274 gst_audioringbuffer_init (GstAudioRingBuffer * ringbuffer, |
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275 GstAudioRingBufferClass * g_class) |
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276 { |
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277 ringbuffer->running = FALSE; |
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278 ringbuffer->queuedseg = 0; |
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279 |
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280 ringbuffer->cond = g_cond_new (); |
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281 } |
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282 |
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283 static void |
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284 gst_audioringbuffer_dispose (GObject * object) |
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285 { |
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286 GstAudioRingBuffer *ringbuffer = GST_AUDIORING_BUFFER (object); |
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287 |
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288 if (ringbuffer->cond) { |
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289 g_cond_free (ringbuffer->cond); |
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290 ringbuffer->cond = NULL; |
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291 } |
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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 G_OBJECT_CLASS (ring_parent_class)->finalize (object); |
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300 } |
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301 |
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302 static gboolean |
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303 gst_audioringbuffer_open_device (GstRingBuffer * buf) |
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304 { |
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305 GstAudioSrc *src; |
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306 GstAudioSrcClass *csrc; |
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307 gboolean result = TRUE; |
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308 |
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309 src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf)); |
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310 csrc = GST_AUDIO_SRC_GET_CLASS (src); |
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311 |
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312 if (csrc->open) |
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313 result = csrc->open (src); |
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314 |
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315 if (!result) |
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316 goto could_not_open; |
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317 |
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318 return result; |
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319 |
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320 could_not_open: |
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321 { |
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322 return FALSE; |
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323 } |
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324 } |
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325 |
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326 static gboolean |
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327 gst_audioringbuffer_close_device (GstRingBuffer * buf) |
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328 { |
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329 GstAudioSrc *src; |
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330 GstAudioSrcClass *csrc; |
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331 gboolean result = TRUE; |
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332 |
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333 src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf)); |
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334 csrc = GST_AUDIO_SRC_GET_CLASS (src); |
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335 |
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336 if (csrc->close) |
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337 result = csrc->close (src); |
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338 |
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339 if (!result) |
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340 goto could_not_open; |
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341 |
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342 return result; |
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343 |
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344 could_not_open: |
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345 { |
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346 return FALSE; |
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347 } |
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348 } |
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349 |
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350 static gboolean |
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351 gst_audioringbuffer_acquire (GstRingBuffer * buf, GstRingBufferSpec * spec) |
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352 { |
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353 GstAudioSrc *src; |
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354 GstAudioSrcClass *csrc; |
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355 GstAudioRingBuffer *abuf; |
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356 gboolean result = FALSE; |
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357 |
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358 src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf)); |
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359 csrc = GST_AUDIO_SRC_GET_CLASS (src); |
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360 |
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361 if (csrc->prepare) |
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362 result = csrc->prepare (src, spec); |
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363 |
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364 if (!result) |
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365 goto could_not_open; |
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366 |
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367 /* allocate one more segment as we need some headroom */ |
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368 spec->segtotal++; |
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369 |
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370 buf->data = gst_buffer_new_and_alloc (spec->segtotal * spec->segsize); |
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371 memset (GST_BUFFER_DATA (buf->data), 0, GST_BUFFER_SIZE (buf->data)); |
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372 |
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373 abuf = GST_AUDIORING_BUFFER (buf); |
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374 abuf->running = TRUE; |
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375 |
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376 src->thread = |
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377 g_thread_create ((GThreadFunc) audioringbuffer_thread_func, buf, TRUE, |
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378 NULL); |
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379 GST_AUDIORING_BUFFER_WAIT (buf); |
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380 |
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381 return result; |
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382 |
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383 could_not_open: |
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384 { |
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385 return FALSE; |
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386 } |
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387 } |
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388 |
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389 /* function is called with LOCK */ |
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390 static gboolean |
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391 gst_audioringbuffer_release (GstRingBuffer * buf) |
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392 { |
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393 GstAudioSrc *src; |
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394 GstAudioSrcClass *csrc; |
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395 GstAudioRingBuffer *abuf; |
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396 gboolean result = FALSE; |
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397 |
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398 src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf)); |
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399 csrc = GST_AUDIO_SRC_GET_CLASS (src); |
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400 abuf = GST_AUDIORING_BUFFER (buf); |
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401 |
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402 abuf->running = FALSE; |
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403 GST_AUDIORING_BUFFER_SIGNAL (buf); |
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404 GST_OBJECT_UNLOCK (buf); |
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405 |
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406 /* join the thread */ |
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407 g_thread_join (src->thread); |
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408 |
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409 GST_OBJECT_LOCK (buf); |
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410 |
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411 /* free the buffer */ |
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412 gst_buffer_unref (buf->data); |
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413 buf->data = NULL; |
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414 |
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415 if (csrc->unprepare) |
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416 result = csrc->unprepare (src); |
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417 |
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418 return result; |
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419 } |
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420 |
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421 static gboolean |
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422 gst_audioringbuffer_start (GstRingBuffer * buf) |
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423 { |
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424 GstAudioSrc *src; |
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425 |
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426 src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf)); |
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427 |
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428 GST_DEBUG ("start, sending signal"); |
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429 GST_AUDIORING_BUFFER_SIGNAL (buf); |
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430 |
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431 return TRUE; |
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432 } |
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433 |
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434 static gboolean |
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435 gst_audioringbuffer_stop (GstRingBuffer * buf) |
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436 { |
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437 GstAudioSrc *src; |
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438 GstAudioSrcClass *csrc; |
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439 |
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440 src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf)); |
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441 csrc = GST_AUDIO_SRC_GET_CLASS (src); |
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442 |
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443 /* unblock any pending writes to the audio device */ |
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444 if (csrc->reset) { |
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445 GST_DEBUG ("reset..."); |
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446 csrc->reset (src); |
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447 GST_DEBUG ("reset done"); |
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448 } |
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449 |
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450 GST_DEBUG ("stop, waiting..."); |
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451 GST_AUDIORING_BUFFER_WAIT (buf); |
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452 GST_DEBUG ("stoped"); |
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453 |
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454 return TRUE; |
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455 } |
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456 |
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457 static guint |
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458 gst_audioringbuffer_delay (GstRingBuffer * buf) |
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459 { |
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460 GstAudioSrc *src; |
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461 GstAudioSrcClass *csrc; |
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462 guint res = 0; |
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463 |
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464 src = GST_AUDIO_SRC (GST_OBJECT_PARENT (buf)); |
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465 csrc = GST_AUDIO_SRC_GET_CLASS (src); |
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466 |
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467 if (csrc->delay) |
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468 res = csrc->delay (src); |
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469 |
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470 return res; |
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471 } |
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472 |
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473 /* AudioSrc signals and args */ |
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474 enum |
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475 { |
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476 /* FILL ME */ |
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477 LAST_SIGNAL |
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478 }; |
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479 |
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480 enum |
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481 { |
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482 ARG_0, |
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483 }; |
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484 |
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485 #define _do_init(bla) \ |
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486 GST_DEBUG_CATEGORY_INIT (gst_audio_src_debug, "audiosrc", 0, "audiosrc element"); |
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487 |
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488 GST_BOILERPLATE_FULL (GstAudioSrc, gst_audio_src, GstBaseAudioSrc, |
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489 GST_TYPE_BASE_AUDIO_SRC, _do_init); |
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490 |
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491 static GstRingBuffer *gst_audio_src_create_ringbuffer (GstBaseAudioSrc * src); |
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492 |
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493 static void |
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494 gst_audio_src_base_init (gpointer g_class) |
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495 { |
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496 } |
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497 |
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498 static void |
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499 gst_audio_src_class_init (GstAudioSrcClass * klass) |
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500 { |
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501 GObjectClass *gobject_class; |
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502 GstElementClass *gstelement_class; |
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503 GstBaseSrcClass *gstbasesrc_class; |
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504 GstPushSrcClass *gstpushsrc_class; |
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505 GstBaseAudioSrcClass *gstbaseaudiosrc_class; |
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506 |
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507 gobject_class = (GObjectClass *) klass; |
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508 gstelement_class = (GstElementClass *) klass; |
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509 gstbasesrc_class = (GstBaseSrcClass *) klass; |
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510 gstpushsrc_class = (GstPushSrcClass *) klass; |
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511 gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; |
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512 |
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513 gstbaseaudiosrc_class->create_ringbuffer = |
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514 GST_DEBUG_FUNCPTR (gst_audio_src_create_ringbuffer); |
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515 } |
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516 |
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517 static void |
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518 gst_audio_src_init (GstAudioSrc * audiosrc, GstAudioSrcClass * g_class) |
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519 { |
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520 } |
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521 |
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522 static GstRingBuffer * |
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523 gst_audio_src_create_ringbuffer (GstBaseAudioSrc * src) |
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524 { |
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525 GstRingBuffer *buffer; |
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526 |
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527 GST_DEBUG ("creating ringbuffer"); |
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528 buffer = g_object_new (GST_TYPE_AUDIORING_BUFFER, NULL); |
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529 GST_DEBUG ("created ringbuffer @%p", buffer); |
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530 |
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531 return buffer; |
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532 } |