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1 /* |
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2 * GStreamer |
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3 * Copyright (C) 2009 Sebastian Dröge <sebastian.droege@collabora.co.uk> |
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4 * |
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5 * This library is free software; you can redistribute it and/or |
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6 * modify it under the terms of the GNU Library General Public |
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7 * License as published by the Free Software Foundation; either |
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8 * version 2 of the License, or (at your option) any later version. |
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9 * |
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10 * This library is distributed in the hope that it will be useful, |
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11 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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13 * Library General Public License for more details. |
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14 * |
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15 * You should have received a copy of the GNU Library General Public |
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16 * License along with this library; if not, write to the |
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17 * Free Software Foundation, Inc., 59 Temple Place - Suite 330, |
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18 * Boston, MA 02111-1307, USA. |
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19 */ |
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20 |
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21 /** |
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22 * SECTION:element-audioecho |
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23 * @Since: 0.10.14 |
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24 * |
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25 * audioecho adds an echo or (simple) reverb effect to an audio stream. The echo |
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26 * delay, intensity and the percentage of feedback can be configured. |
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27 * |
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28 * For getting an echo effect you have to set the delay to a larger value, |
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29 * for example 200ms and more. Everything below will result in a simple |
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30 * reverb effect, which results in a slightly metallic sound. |
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31 * |
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32 * Use the max-delay property to set the maximum amount of delay that |
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33 * will be used. This can only be set before going to the PAUSED or PLAYING |
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34 * state and will be set to the current delay by default. |
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35 * |
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36 * <refsect2> |
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37 * <title>Example launch line</title> |
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38 * |[ |
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39 * gst-launch filesrc location="melo1.ogg" ! audioconvert ! audioecho delay=500000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink |
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40 * gst-launch filesrc location="melo1.ogg" ! decodebin ! audioconvert ! audioecho delay=50000000 intensity=0.6 feedback=0.4 ! audioconvert ! autoaudiosink |
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41 * ]| |
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42 * </refsect2> |
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43 */ |
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44 |
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45 #ifdef HAVE_CONFIG_H |
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46 #include "config.h" |
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47 #endif |
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48 |
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49 #include <gst/gst.h> |
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50 #include <gst/base/gstbasetransform.h> |
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51 #include <gst/audio/audio.h> |
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52 #include <gst/audio/gstaudiofilter.h> |
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53 #include <gst/controller/gstcontroller.h> |
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54 |
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55 #include "audioecho.h" |
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56 |
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57 #define GST_CAT_DEFAULT gst_audio_echo_debug |
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58 GST_DEBUG_CATEGORY_STATIC (GST_CAT_DEFAULT); |
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59 |
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60 enum |
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61 { |
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62 PROP_0, |
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63 PROP_DELAY, |
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64 PROP_MAX_DELAY, |
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65 PROP_INTENSITY, |
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66 PROP_FEEDBACK |
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67 }; |
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68 |
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69 #define ALLOWED_CAPS \ |
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70 "audio/x-raw-float," \ |
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71 " width=(int) { 32, 64 }, " \ |
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72 " endianness=(int)BYTE_ORDER," \ |
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73 " rate=(int)[1,MAX]," \ |
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74 " channels=(int)[1,MAX]" |
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75 |
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76 #define DEBUG_INIT(bla) \ |
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77 GST_DEBUG_CATEGORY_INIT (gst_audio_echo_debug, "audioecho", 0, "audioecho element"); |
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78 |
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79 GST_BOILERPLATE_FULL (GstAudioEcho, gst_audio_echo, GstAudioFilter, |
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80 GST_TYPE_AUDIO_FILTER, DEBUG_INIT); |
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81 |
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82 static void gst_audio_echo_set_property (GObject * object, guint prop_id, |
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83 const GValue * value, GParamSpec * pspec); |
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84 static void gst_audio_echo_get_property (GObject * object, guint prop_id, |
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85 GValue * value, GParamSpec * pspec); |
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86 static void gst_audio_echo_finalize (GObject * object); |
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87 |
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88 static gboolean gst_audio_echo_setup (GstAudioFilter * self, |
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89 GstRingBufferSpec * format); |
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90 static gboolean gst_audio_echo_stop (GstBaseTransform * base); |
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91 static GstFlowReturn gst_audio_echo_transform_ip (GstBaseTransform * base, |
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92 GstBuffer * buf); |
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93 |
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94 static void gst_audio_echo_transform_float (GstAudioEcho * self, |
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95 gfloat * data, guint num_samples); |
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96 static void gst_audio_echo_transform_double (GstAudioEcho * self, |
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97 gdouble * data, guint num_samples); |
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98 |
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99 /* GObject vmethod implementations */ |
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100 |
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101 static void |
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102 gst_audio_echo_base_init (gpointer klass) |
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103 { |
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104 GstElementClass *element_class = GST_ELEMENT_CLASS (klass); |
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105 GstCaps *caps; |
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106 |
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107 gst_element_class_set_details_simple (element_class, "Audio echo", |
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108 "Filter/Effect/Audio", |
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109 "Adds an echo or reverb effect to an audio stream", |
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110 "Sebastian Dröge <sebastian.droege@collabora.co.uk>"); |
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111 |
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112 caps = gst_caps_from_string (ALLOWED_CAPS); |
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113 gst_audio_filter_class_add_pad_templates (GST_AUDIO_FILTER_CLASS (klass), |
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114 caps); |
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115 gst_caps_unref (caps); |
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116 } |
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117 |
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118 static void |
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119 gst_audio_echo_class_init (GstAudioEchoClass * klass) |
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120 { |
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121 GObjectClass *gobject_class = (GObjectClass *) klass; |
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122 GstBaseTransformClass *basetransform_class = (GstBaseTransformClass *) klass; |
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123 GstAudioFilterClass *audioself_class = (GstAudioFilterClass *) klass; |
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124 |
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125 gobject_class->set_property = gst_audio_echo_set_property; |
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126 gobject_class->get_property = gst_audio_echo_get_property; |
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127 gobject_class->finalize = gst_audio_echo_finalize; |
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128 |
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129 g_object_class_install_property (gobject_class, PROP_DELAY, |
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130 g_param_spec_uint64 ("delay", "Delay", |
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131 "Delay of the echo in nanoseconds", 1, G_MAXUINT64, |
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132 1, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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133 | GST_PARAM_CONTROLLABLE)); |
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134 |
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135 g_object_class_install_property (gobject_class, PROP_MAX_DELAY, |
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136 g_param_spec_uint64 ("max-delay", "Maximum Delay", |
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137 "Maximum delay of the echo in nanoseconds" |
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138 " (can't be changed in PLAYING or PAUSED state)", |
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139 1, G_MAXUINT64, 1, |
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140 G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | GST_PARAM_CONTROLLABLE)); |
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141 |
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142 g_object_class_install_property (gobject_class, PROP_INTENSITY, |
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143 g_param_spec_float ("intensity", "Intensity", |
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144 "Intensity of the echo", 0.0, 1.0, |
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145 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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146 | GST_PARAM_CONTROLLABLE)); |
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147 |
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148 g_object_class_install_property (gobject_class, PROP_FEEDBACK, |
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149 g_param_spec_float ("feedback", "Feedback", |
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150 "Amount of feedback", 0.0, 1.0, |
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151 0.0, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS |
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152 | GST_PARAM_CONTROLLABLE)); |
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153 |
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154 audioself_class->setup = GST_DEBUG_FUNCPTR (gst_audio_echo_setup); |
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155 basetransform_class->transform_ip = |
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156 GST_DEBUG_FUNCPTR (gst_audio_echo_transform_ip); |
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157 basetransform_class->stop = GST_DEBUG_FUNCPTR (gst_audio_echo_stop); |
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158 } |
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159 |
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160 static void |
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161 gst_audio_echo_init (GstAudioEcho * self, GstAudioEchoClass * klass) |
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162 { |
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163 self->delay = 1; |
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164 self->max_delay = 1; |
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165 self->intensity = 0.0; |
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166 self->feedback = 0.0; |
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167 |
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168 gst_base_transform_set_in_place (GST_BASE_TRANSFORM (self), TRUE); |
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169 } |
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170 |
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171 static void |
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172 gst_audio_echo_finalize (GObject * object) |
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173 { |
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174 GstAudioEcho *self = GST_AUDIO_ECHO (object); |
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175 |
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176 g_free (self->buffer); |
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177 self->buffer = NULL; |
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178 |
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179 G_OBJECT_CLASS (parent_class)->finalize (object); |
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180 } |
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181 |
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182 static void |
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183 gst_audio_echo_set_property (GObject * object, guint prop_id, |
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184 const GValue * value, GParamSpec * pspec) |
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185 { |
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186 GstAudioEcho *self = GST_AUDIO_ECHO (object); |
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187 |
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188 switch (prop_id) { |
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189 case PROP_DELAY:{ |
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190 guint64 max_delay, delay; |
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191 |
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192 GST_BASE_TRANSFORM_LOCK (self); |
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193 delay = g_value_get_uint64 (value); |
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194 max_delay = self->max_delay; |
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195 |
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196 if (delay > max_delay && GST_STATE (self) > GST_STATE_READY) { |
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197 GST_WARNING_OBJECT (self, "New delay (%" GST_TIME_FORMAT ") " |
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198 "is larger than maximum delay (%" GST_TIME_FORMAT ")", |
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199 GST_TIME_ARGS (delay), GST_TIME_ARGS (max_delay)); |
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200 self->delay = max_delay; |
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201 } else { |
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202 self->delay = delay; |
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203 self->max_delay = MAX (delay, max_delay); |
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204 } |
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205 GST_BASE_TRANSFORM_UNLOCK (self); |
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206 } |
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207 break; |
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208 case PROP_MAX_DELAY:{ |
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209 guint64 max_delay, delay; |
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210 |
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211 GST_BASE_TRANSFORM_LOCK (self); |
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212 max_delay = g_value_get_uint64 (value); |
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213 delay = self->delay; |
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214 |
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215 if (GST_STATE (self) > GST_STATE_READY) { |
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216 GST_ERROR_OBJECT (self, "Can't change maximum delay in" |
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217 " PLAYING or PAUSED state"); |
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218 } else { |
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219 self->delay = delay; |
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220 self->max_delay = max_delay; |
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221 } |
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222 GST_BASE_TRANSFORM_UNLOCK (self); |
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223 } |
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224 break; |
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225 case PROP_INTENSITY:{ |
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226 GST_BASE_TRANSFORM_LOCK (self); |
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227 self->intensity = g_value_get_float (value); |
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228 GST_BASE_TRANSFORM_UNLOCK (self); |
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229 } |
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230 break; |
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231 case PROP_FEEDBACK:{ |
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232 GST_BASE_TRANSFORM_LOCK (self); |
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233 self->feedback = g_value_get_float (value); |
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234 GST_BASE_TRANSFORM_UNLOCK (self); |
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235 } |
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236 break; |
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237 default: |
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238 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
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239 break; |
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240 } |
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241 } |
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242 |
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243 static void |
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244 gst_audio_echo_get_property (GObject * object, guint prop_id, |
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245 GValue * value, GParamSpec * pspec) |
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246 { |
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247 GstAudioEcho *self = GST_AUDIO_ECHO (object); |
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248 |
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249 switch (prop_id) { |
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250 case PROP_DELAY: |
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251 GST_BASE_TRANSFORM_LOCK (self); |
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252 g_value_set_uint64 (value, self->delay); |
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253 GST_BASE_TRANSFORM_UNLOCK (self); |
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254 break; |
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255 case PROP_MAX_DELAY: |
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256 GST_BASE_TRANSFORM_LOCK (self); |
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257 g_value_set_uint64 (value, self->max_delay); |
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258 GST_BASE_TRANSFORM_UNLOCK (self); |
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259 break; |
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260 case PROP_INTENSITY: |
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261 GST_BASE_TRANSFORM_LOCK (self); |
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262 g_value_set_float (value, self->intensity); |
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263 GST_BASE_TRANSFORM_UNLOCK (self); |
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264 break; |
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265 case PROP_FEEDBACK: |
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266 GST_BASE_TRANSFORM_LOCK (self); |
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267 g_value_set_float (value, self->feedback); |
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268 GST_BASE_TRANSFORM_UNLOCK (self); |
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269 break; |
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270 default: |
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271 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
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272 break; |
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273 } |
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274 } |
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275 |
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276 /* GstAudioFilter vmethod implementations */ |
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277 |
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278 static gboolean |
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279 gst_audio_echo_setup (GstAudioFilter * base, GstRingBufferSpec * format) |
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280 { |
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281 GstAudioEcho *self = GST_AUDIO_ECHO (base); |
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282 gboolean ret = TRUE; |
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283 |
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284 if (format->type == GST_BUFTYPE_FLOAT && format->width == 32) |
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285 self->process = (GstAudioEchoProcessFunc) |
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286 gst_audio_echo_transform_float; |
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287 else if (format->type == GST_BUFTYPE_FLOAT && format->width == 64) |
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288 self->process = (GstAudioEchoProcessFunc) |
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289 gst_audio_echo_transform_double; |
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290 else |
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291 ret = FALSE; |
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292 |
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293 g_free (self->buffer); |
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294 self->buffer = NULL; |
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295 self->buffer_pos = 0; |
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296 self->buffer_size = 0; |
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297 self->buffer_size_frames = 0; |
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298 |
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299 return ret; |
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300 } |
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301 |
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302 static gboolean |
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303 gst_audio_echo_stop (GstBaseTransform * base) |
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304 { |
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305 GstAudioEcho *self = GST_AUDIO_ECHO (base); |
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306 |
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307 g_free (self->buffer); |
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308 self->buffer = NULL; |
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309 self->buffer_pos = 0; |
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310 self->buffer_size = 0; |
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311 self->buffer_size_frames = 0; |
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312 |
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313 return TRUE; |
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314 } |
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315 |
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316 #define TRANSFORM_FUNC(name, type) \ |
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317 static void \ |
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318 gst_audio_echo_transform_##name (GstAudioEcho * self, \ |
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319 type * data, guint num_samples) \ |
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320 { \ |
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321 type *buffer = (type *) self->buffer; \ |
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322 guint channels = GST_AUDIO_FILTER (self)->format.channels; \ |
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323 guint rate = GST_AUDIO_FILTER (self)->format.rate; \ |
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324 guint i, j; \ |
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325 guint echo_index = self->buffer_size_frames - self->delay_frames; \ |
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326 gdouble echo_off = ((((gdouble) self->delay) * rate) / GST_SECOND) - self->delay_frames; \ |
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327 \ |
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328 if (echo_off < 0.0) \ |
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329 echo_off = 0.0; \ |
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330 \ |
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331 num_samples /= channels; \ |
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332 \ |
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333 for (i = 0; i < num_samples; i++) { \ |
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334 guint echo0_index = ((echo_index + self->buffer_pos) % self->buffer_size_frames) * channels; \ |
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335 guint echo1_index = ((echo_index + self->buffer_pos +1) % self->buffer_size_frames) * channels; \ |
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336 guint rbout_index = (self->buffer_pos % self->buffer_size_frames) * channels; \ |
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337 for (j = 0; j < channels; j++) { \ |
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338 gdouble in = data[i*channels + j]; \ |
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339 gdouble echo0 = buffer[echo0_index + j]; \ |
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340 gdouble echo1 = buffer[echo1_index + j]; \ |
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341 gdouble echo = echo0 + (echo1-echo0)*echo_off; \ |
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342 type out = in + self->intensity * echo; \ |
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343 \ |
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344 data[i*channels + j] = out; \ |
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345 \ |
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346 buffer[rbout_index + j] = in + self->feedback * echo; \ |
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347 } \ |
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348 self->buffer_pos = (self->buffer_pos + 1) % self->buffer_size_frames; \ |
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349 } \ |
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350 } |
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351 |
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352 TRANSFORM_FUNC (float, gfloat); |
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353 TRANSFORM_FUNC (double, gdouble); |
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354 |
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355 /* GstBaseTransform vmethod implementations */ |
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356 static GstFlowReturn |
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357 gst_audio_echo_transform_ip (GstBaseTransform * base, GstBuffer * buf) |
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358 { |
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359 GstAudioEcho *self = GST_AUDIO_ECHO (base); |
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360 guint num_samples = |
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361 GST_BUFFER_SIZE (buf) / (GST_AUDIO_FILTER (self)->format.width / 8); |
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362 |
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363 if (GST_CLOCK_TIME_IS_VALID (GST_BUFFER_TIMESTAMP (buf))) |
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364 gst_object_sync_values (G_OBJECT (self), GST_BUFFER_TIMESTAMP (buf)); |
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365 |
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366 if (self->buffer == NULL) { |
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367 guint width, rate, channels; |
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368 |
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369 width = GST_AUDIO_FILTER (self)->format.width / 8; |
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370 rate = GST_AUDIO_FILTER (self)->format.rate; |
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371 channels = GST_AUDIO_FILTER (self)->format.channels; |
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372 |
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373 self->delay_frames = |
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374 MAX (gst_util_uint64_scale (self->delay, rate, GST_SECOND), 1); |
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375 self->buffer_size_frames = |
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376 MAX (gst_util_uint64_scale (self->max_delay, rate, GST_SECOND), 1); |
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377 |
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378 self->buffer_size = self->buffer_size_frames * width * channels; |
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379 self->buffer = g_try_malloc0 (self->buffer_size); |
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380 self->buffer_pos = 0; |
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381 |
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382 if (self->buffer == NULL) { |
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383 GST_ERROR_OBJECT (self, "Failed to allocate %u bytes", self->buffer_size); |
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384 return GST_FLOW_ERROR; |
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385 } |
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386 } |
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387 |
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388 self->process (self, GST_BUFFER_DATA (buf), num_samples); |
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389 |
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390 return GST_FLOW_OK; |
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391 } |