1 /* GStreamer |
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2 * Copyright (C) 2005 Wim Taymans <wim@fluendo.com> |
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3 * |
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4 * gstalsasrc.c: |
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5 * |
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6 * This library is free software; you can redistribute it and/or |
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7 * modify it under the terms of the GNU Library General Public |
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8 * License as published by the Free Software Foundation; either |
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9 * version 2 of the License, or (at your option) any later version. |
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10 * |
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11 * This library is distributed in the hope that it will be useful, |
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12 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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13 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU |
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14 * Library General Public License for more details. |
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15 * |
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16 * You should have received a copy of the GNU Library General Public |
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17 * License along with this library; if not, write to the |
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18 * Free Software Foundation, Inc., 59 Temple Place - Suite 330, |
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19 * Boston, MA 02111-1307, USA. |
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20 */ |
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21 |
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22 /** |
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23 * SECTION:element-alsasrc |
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24 * @short_description: capture audio from an alsa device |
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25 * @see_also: alsasink, alsamixer |
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26 * |
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27 * <refsect2> |
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28 * <para> |
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29 * This element reads data from an audio card using the ALSA API. |
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30 * </para> |
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31 * <title>Example pipelines</title> |
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32 * <para> |
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33 * Record from a sound card using ALSA and encode to Ogg/Vorbis. |
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34 * </para> |
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35 * <programlisting> |
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36 * gst-launch -v alsasrc ! audioconvert ! vorbisenc ! oggmux ! filesink location=alsasrc.ogg |
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37 * </programlisting> |
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38 * </refsect2> |
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39 * |
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40 * Last reviewed on 2006-03-01 (0.10.4) |
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41 */ |
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42 |
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43 #ifdef HAVE_CONFIG_H |
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44 #include "config.h" |
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45 #endif |
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46 #include <sys/ioctl.h> |
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47 #include <fcntl.h> |
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48 #include <errno.h> |
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49 #include <unistd.h> |
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50 #include <string.h> |
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51 #include <getopt.h> |
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52 #include <alsa/asoundlib.h> |
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53 |
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54 #include "gstalsasrc.h" |
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55 #include "gstalsadeviceprobe.h" |
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56 |
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57 #include <gst/gst-i18n-plugin.h> |
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58 |
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59 /* elementfactory information */ |
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60 static const GstElementDetails gst_alsasrc_details = |
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61 GST_ELEMENT_DETAILS ("Audio source (ALSA)", |
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62 "Source/Audio", |
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63 "Read from a sound card via ALSA", |
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64 "Wim Taymans <wim@fluendo.com>"); |
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65 |
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66 #define DEFAULT_PROP_DEVICE "default" |
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67 #define DEFAULT_PROP_DEVICE_NAME "" |
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68 |
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69 enum |
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70 { |
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71 PROP_0, |
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72 PROP_DEVICE, |
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73 PROP_DEVICE_NAME, |
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74 }; |
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75 |
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76 static void gst_alsasrc_init_interfaces (GType type); |
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77 |
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78 GST_BOILERPLATE_FULL (GstAlsaSrc, gst_alsasrc, GstAudioSrc, |
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79 GST_TYPE_AUDIO_SRC, gst_alsasrc_init_interfaces); |
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80 |
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81 GST_IMPLEMENT_ALSA_MIXER_METHODS (GstAlsaSrc, gst_alsasrc_mixer); |
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82 |
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83 static void gst_alsasrc_finalize (GObject * object); |
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84 static void gst_alsasrc_set_property (GObject * object, |
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85 guint prop_id, const GValue * value, GParamSpec * pspec); |
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86 static void gst_alsasrc_get_property (GObject * object, |
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87 guint prop_id, GValue * value, GParamSpec * pspec); |
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88 |
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89 static GstCaps *gst_alsasrc_getcaps (GstBaseSrc * bsrc); |
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90 |
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91 static gboolean gst_alsasrc_open (GstAudioSrc * asrc); |
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92 static gboolean gst_alsasrc_prepare (GstAudioSrc * asrc, |
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93 GstRingBufferSpec * spec); |
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94 static gboolean gst_alsasrc_unprepare (GstAudioSrc * asrc); |
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95 static gboolean gst_alsasrc_close (GstAudioSrc * asrc); |
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96 static guint gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length); |
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97 static guint gst_alsasrc_delay (GstAudioSrc * asrc); |
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98 static void gst_alsasrc_reset (GstAudioSrc * asrc); |
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99 |
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100 /* AlsaSrc signals and args */ |
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101 enum |
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102 { |
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103 LAST_SIGNAL |
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104 }; |
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105 |
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106 #if (G_BYTE_ORDER == G_LITTLE_ENDIAN) |
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107 # define ALSA_SRC_FACTORY_ENDIANNESS "LITTLE_ENDIAN, BIG_ENDIAN" |
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108 #else |
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109 # define ALSA_SRC_FACTORY_ENDIANNESS "BIG_ENDIAN, LITTLE_ENDIAN" |
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110 #endif |
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111 |
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112 static GstStaticPadTemplate alsasrc_src_factory = |
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113 GST_STATIC_PAD_TEMPLATE ("src", |
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114 GST_PAD_SRC, |
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115 GST_PAD_ALWAYS, |
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116 GST_STATIC_CAPS ("audio/x-raw-int, " |
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117 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, " |
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118 "signed = (boolean) { TRUE, FALSE }, " |
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119 "width = (int) 32, " |
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120 "depth = (int) 32, " |
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121 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " |
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122 "audio/x-raw-int, " |
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123 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, " |
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124 "signed = (boolean) { TRUE, FALSE }, " |
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125 "width = (int) 32, " |
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126 "depth = (int) 24, " |
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127 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " |
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128 "audio/x-raw-int, " |
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129 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, " |
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130 "signed = (boolean) { TRUE, FALSE }, " |
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131 "width = (int) 24, " |
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132 "depth = (int) 24, " |
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133 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " |
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134 "audio/x-raw-int, " |
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135 "endianness = (int) { " ALSA_SRC_FACTORY_ENDIANNESS " }, " |
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136 "signed = (boolean) { TRUE, FALSE }, " |
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137 "width = (int) 16, " |
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138 "depth = (int) 16, " |
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139 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]; " |
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140 "audio/x-raw-int, " |
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141 "signed = (boolean) { TRUE, FALSE }, " |
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142 "width = (int) 8, " |
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143 "depth = (int) 8, " |
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144 "rate = (int) [ 1, MAX ], " "channels = (int) [ 1, MAX ]") |
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145 ); |
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146 |
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147 static void |
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148 gst_alsasrc_finalize (GObject * object) |
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149 { |
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150 GstAlsaSrc *src = GST_ALSA_SRC (object); |
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151 |
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152 g_free (src->device); |
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153 g_mutex_free (src->alsa_lock); |
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154 |
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155 G_OBJECT_CLASS (parent_class)->finalize (object); |
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156 } |
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157 |
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158 static gboolean |
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159 gst_alsasrc_interface_supported (GstAlsaSrc * this, GType interface_type) |
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160 { |
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161 /* only support this one interface (wrapped by GstImplementsInterface) */ |
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162 g_assert (interface_type == GST_TYPE_MIXER); |
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163 |
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164 return gst_alsasrc_mixer_supported (this, interface_type); |
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165 } |
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166 |
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167 static void |
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168 gst_implements_interface_init (GstImplementsInterfaceClass * klass) |
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169 { |
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170 klass->supported = (gpointer) gst_alsasrc_interface_supported; |
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171 } |
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172 |
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173 static void |
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174 gst_alsasrc_init_interfaces (GType type) |
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175 { |
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176 static const GInterfaceInfo implements_iface_info = { |
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177 (GInterfaceInitFunc) gst_implements_interface_init, |
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178 NULL, |
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179 NULL, |
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180 }; |
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181 static const GInterfaceInfo mixer_iface_info = { |
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182 (GInterfaceInitFunc) gst_alsasrc_mixer_interface_init, |
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183 NULL, |
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184 NULL, |
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185 }; |
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186 |
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187 g_type_add_interface_static (type, GST_TYPE_IMPLEMENTS_INTERFACE, |
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188 &implements_iface_info); |
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189 g_type_add_interface_static (type, GST_TYPE_MIXER, &mixer_iface_info); |
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190 |
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191 gst_alsa_type_add_device_property_probe_interface (type); |
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192 } |
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193 |
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194 static void |
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195 gst_alsasrc_base_init (gpointer g_class) |
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196 { |
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197 GstElementClass *element_class = GST_ELEMENT_CLASS (g_class); |
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198 |
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199 gst_element_class_set_details (element_class, &gst_alsasrc_details); |
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200 |
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201 gst_element_class_add_pad_template (element_class, |
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202 gst_static_pad_template_get (&alsasrc_src_factory)); |
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203 } |
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204 |
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205 static void |
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206 gst_alsasrc_class_init (GstAlsaSrcClass * klass) |
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207 { |
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208 GObjectClass *gobject_class; |
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209 GstElementClass *gstelement_class; |
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210 GstBaseSrcClass *gstbasesrc_class; |
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211 GstBaseAudioSrcClass *gstbaseaudiosrc_class; |
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212 GstAudioSrcClass *gstaudiosrc_class; |
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213 |
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214 gobject_class = (GObjectClass *) klass; |
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215 gstelement_class = (GstElementClass *) klass; |
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216 gstbasesrc_class = (GstBaseSrcClass *) klass; |
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217 gstbaseaudiosrc_class = (GstBaseAudioSrcClass *) klass; |
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218 gstaudiosrc_class = (GstAudioSrcClass *) klass; |
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219 |
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220 gobject_class->finalize = GST_DEBUG_FUNCPTR (gst_alsasrc_finalize); |
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221 gobject_class->get_property = GST_DEBUG_FUNCPTR (gst_alsasrc_get_property); |
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222 gobject_class->set_property = GST_DEBUG_FUNCPTR (gst_alsasrc_set_property); |
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223 |
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224 gstbasesrc_class->get_caps = GST_DEBUG_FUNCPTR (gst_alsasrc_getcaps); |
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225 |
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226 gstaudiosrc_class->open = GST_DEBUG_FUNCPTR (gst_alsasrc_open); |
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227 gstaudiosrc_class->prepare = GST_DEBUG_FUNCPTR (gst_alsasrc_prepare); |
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228 gstaudiosrc_class->unprepare = GST_DEBUG_FUNCPTR (gst_alsasrc_unprepare); |
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229 gstaudiosrc_class->close = GST_DEBUG_FUNCPTR (gst_alsasrc_close); |
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230 gstaudiosrc_class->read = GST_DEBUG_FUNCPTR (gst_alsasrc_read); |
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231 gstaudiosrc_class->delay = GST_DEBUG_FUNCPTR (gst_alsasrc_delay); |
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232 gstaudiosrc_class->reset = GST_DEBUG_FUNCPTR (gst_alsasrc_reset); |
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233 |
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234 g_object_class_install_property (gobject_class, PROP_DEVICE, |
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235 g_param_spec_string ("device", "Device", |
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236 "ALSA device, as defined in an asound configuration file", |
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237 DEFAULT_PROP_DEVICE, G_PARAM_READWRITE)); |
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238 |
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239 g_object_class_install_property (gobject_class, PROP_DEVICE_NAME, |
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240 g_param_spec_string ("device-name", "Device name", |
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241 "Human-readable name of the sound device", |
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242 DEFAULT_PROP_DEVICE_NAME, G_PARAM_READABLE)); |
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243 } |
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244 |
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245 static void |
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246 gst_alsasrc_set_property (GObject * object, guint prop_id, |
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247 const GValue * value, GParamSpec * pspec) |
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248 { |
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249 GstAlsaSrc *src; |
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250 |
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251 src = GST_ALSA_SRC (object); |
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252 |
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253 switch (prop_id) { |
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254 case PROP_DEVICE: |
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255 g_free (src->device); |
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256 src->device = g_value_dup_string (value); |
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257 if (src->device == NULL) { |
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258 src->device = g_strdup (DEFAULT_PROP_DEVICE); |
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259 } |
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260 break; |
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261 default: |
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262 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
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263 break; |
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264 } |
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265 } |
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266 |
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267 static void |
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268 gst_alsasrc_get_property (GObject * object, guint prop_id, |
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269 GValue * value, GParamSpec * pspec) |
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270 { |
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271 GstAlsaSrc *src; |
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272 |
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273 src = GST_ALSA_SRC (object); |
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274 |
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275 switch (prop_id) { |
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276 case PROP_DEVICE: |
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277 g_value_set_string (value, src->device); |
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278 break; |
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279 case PROP_DEVICE_NAME: |
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280 g_value_take_string (value, |
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281 gst_alsa_find_device_name (GST_OBJECT_CAST (src), |
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282 src->device, src->handle, SND_PCM_STREAM_CAPTURE)); |
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283 break; |
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284 default: |
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285 G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); |
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286 break; |
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287 } |
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288 } |
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289 |
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290 static void |
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291 gst_alsasrc_init (GstAlsaSrc * alsasrc, GstAlsaSrcClass * g_class) |
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292 { |
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293 GST_DEBUG_OBJECT (alsasrc, "initializing"); |
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294 |
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295 alsasrc->device = g_strdup (DEFAULT_PROP_DEVICE); |
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296 alsasrc->cached_caps = NULL; |
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297 |
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298 alsasrc->alsa_lock = g_mutex_new (); |
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299 } |
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300 |
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301 #define CHECK(call, error) \ |
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302 G_STMT_START { \ |
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303 if ((err = call) < 0) \ |
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304 goto error; \ |
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305 } G_STMT_END; |
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306 |
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307 |
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308 static GstCaps * |
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309 gst_alsasrc_getcaps (GstBaseSrc * bsrc) |
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310 { |
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311 GstElementClass *element_class; |
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312 GstPadTemplate *pad_template; |
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313 GstAlsaSrc *src; |
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314 GstCaps *caps; |
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315 |
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316 src = GST_ALSA_SRC (bsrc); |
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317 |
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318 if (src->handle == NULL) { |
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319 GST_DEBUG_OBJECT (src, "device not open, using template caps"); |
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320 return NULL; /* base class will get template caps for us */ |
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321 } |
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322 |
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323 if (src->cached_caps) { |
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324 GST_LOG_OBJECT (src, "Returning cached caps"); |
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325 return gst_caps_ref (src->cached_caps); |
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326 } |
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327 |
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328 element_class = GST_ELEMENT_GET_CLASS (src); |
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329 pad_template = gst_element_class_get_pad_template (element_class, "src"); |
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330 g_return_val_if_fail (pad_template != NULL, NULL); |
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331 |
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332 caps = gst_alsa_probe_supported_formats (GST_OBJECT (src), src->handle, |
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333 gst_pad_template_get_caps (pad_template)); |
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334 |
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335 if (caps) { |
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336 src->cached_caps = gst_caps_ref (caps); |
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337 } |
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338 |
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339 GST_INFO_OBJECT (src, "returning caps %" GST_PTR_FORMAT, caps); |
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340 |
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341 return caps; |
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342 } |
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343 |
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344 static int |
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345 set_hwparams (GstAlsaSrc * alsa) |
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346 { |
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347 guint rrate; |
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348 gint err, dir; |
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349 snd_pcm_hw_params_t *params; |
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350 |
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351 snd_pcm_hw_params_malloc (¶ms); |
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352 |
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353 /* choose all parameters */ |
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354 CHECK (snd_pcm_hw_params_any (alsa->handle, params), no_config); |
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355 /* set the interleaved read/write format */ |
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356 CHECK (snd_pcm_hw_params_set_access (alsa->handle, params, alsa->access), |
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357 wrong_access); |
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358 /* set the sample format */ |
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359 CHECK (snd_pcm_hw_params_set_format (alsa->handle, params, alsa->format), |
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360 no_sample_format); |
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361 /* set the count of channels */ |
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362 CHECK (snd_pcm_hw_params_set_channels (alsa->handle, params, alsa->channels), |
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363 no_channels); |
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364 /* set the stream rate */ |
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365 rrate = alsa->rate; |
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366 CHECK (snd_pcm_hw_params_set_rate_near (alsa->handle, params, &rrate, NULL), |
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367 no_rate); |
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368 if (rrate != alsa->rate) |
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369 goto rate_match; |
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370 |
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371 if (alsa->buffer_time != -1) { |
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372 /* set the buffer time */ |
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373 CHECK (snd_pcm_hw_params_set_buffer_time_near (alsa->handle, params, |
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374 &alsa->buffer_time, &dir), buffer_time); |
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375 } |
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376 if (alsa->period_time != -1) { |
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377 /* set the period time */ |
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378 CHECK (snd_pcm_hw_params_set_period_time_near (alsa->handle, params, |
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379 &alsa->period_time, &dir), period_time); |
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380 } |
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381 |
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382 /* write the parameters to device */ |
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383 CHECK (snd_pcm_hw_params (alsa->handle, params), set_hw_params); |
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384 |
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385 CHECK (snd_pcm_hw_params_get_buffer_size (params, &alsa->buffer_size), |
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386 buffer_size); |
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387 |
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388 CHECK (snd_pcm_hw_params_get_period_size (params, &alsa->period_size, &dir), |
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389 period_size); |
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390 |
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391 snd_pcm_hw_params_free (params); |
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392 return 0; |
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393 |
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394 /* ERRORS */ |
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395 no_config: |
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396 { |
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397 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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398 ("Broken configuration for recording: no configurations available: %s", |
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399 snd_strerror (err))); |
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400 snd_pcm_hw_params_free (params); |
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401 return err; |
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402 } |
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403 wrong_access: |
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404 { |
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405 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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406 ("Access type not available for recording: %s", snd_strerror (err))); |
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407 snd_pcm_hw_params_free (params); |
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408 return err; |
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409 } |
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410 no_sample_format: |
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411 { |
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412 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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413 ("Sample format not available for recording: %s", snd_strerror (err))); |
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414 snd_pcm_hw_params_free (params); |
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415 return err; |
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416 } |
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417 no_channels: |
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418 { |
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419 gchar *msg = NULL; |
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420 |
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421 if ((alsa->channels) == 1) |
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422 msg = g_strdup (_("Could not open device for recording in mono mode.")); |
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423 if ((alsa->channels) == 2) |
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424 msg = g_strdup (_("Could not open device for recording in stereo mode.")); |
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425 if ((alsa->channels) > 2) |
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426 msg = |
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427 g_strdup_printf (_ |
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428 ("Could not open device for recording in %d-channel mode"), |
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429 alsa->channels); |
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430 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (msg), (snd_strerror (err))); |
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431 g_free (msg); |
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432 snd_pcm_hw_params_free (params); |
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433 return err; |
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434 } |
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435 no_rate: |
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436 { |
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437 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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438 ("Rate %iHz not available for recording: %s", |
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439 alsa->rate, snd_strerror (err))); |
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440 snd_pcm_hw_params_free (params); |
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441 return err; |
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442 } |
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443 rate_match: |
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444 { |
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445 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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446 ("Rate doesn't match (requested %iHz, get %iHz)", alsa->rate, err)); |
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447 snd_pcm_hw_params_free (params); |
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448 return -EINVAL; |
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449 } |
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450 buffer_time: |
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451 { |
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452 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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453 ("Unable to set buffer time %i for recording: %s", |
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454 alsa->buffer_time, snd_strerror (err))); |
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455 snd_pcm_hw_params_free (params); |
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456 return err; |
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457 } |
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458 buffer_size: |
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459 { |
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460 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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461 ("Unable to get buffer size for recording: %s", snd_strerror (err))); |
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462 snd_pcm_hw_params_free (params); |
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463 return err; |
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464 } |
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465 period_time: |
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466 { |
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467 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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468 ("Unable to set period time %i for recording: %s", alsa->period_time, |
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469 snd_strerror (err))); |
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470 snd_pcm_hw_params_free (params); |
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471 return err; |
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472 } |
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473 period_size: |
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474 { |
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475 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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476 ("Unable to get period size for recording: %s", snd_strerror (err))); |
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477 snd_pcm_hw_params_free (params); |
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478 return err; |
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479 } |
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480 set_hw_params: |
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481 { |
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482 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
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483 ("Unable to set hw params for recording: %s", snd_strerror (err))); |
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484 snd_pcm_hw_params_free (params); |
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485 return err; |
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486 } |
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487 } |
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488 |
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489 static int |
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490 set_swparams (GstAlsaSrc * alsa) |
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491 { |
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492 int err; |
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493 snd_pcm_sw_params_t *params; |
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494 |
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495 snd_pcm_sw_params_malloc (¶ms); |
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496 |
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497 /* get the current swparams */ |
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498 CHECK (snd_pcm_sw_params_current (alsa->handle, params), no_config); |
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499 /* allow the transfer when at least period_size samples can be processed */ |
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500 CHECK (snd_pcm_sw_params_set_avail_min (alsa->handle, params, |
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501 alsa->period_size), set_avail); |
|
502 /* start the transfer on first read */ |
|
503 CHECK (snd_pcm_sw_params_set_start_threshold (alsa->handle, params, |
|
504 0), start_threshold); |
|
505 |
|
506 #if GST_CHECK_ALSA_VERSION(1,0,16) |
|
507 /* snd_pcm_sw_params_set_xfer_align() is deprecated, alignment is always 1 */ |
|
508 #else |
|
509 /* align all transfers to 1 sample */ |
|
510 CHECK (snd_pcm_sw_params_set_xfer_align (alsa->handle, params, 1), set_align); |
|
511 #endif |
|
512 |
|
513 /* write the parameters to the recording device */ |
|
514 CHECK (snd_pcm_sw_params (alsa->handle, params), set_sw_params); |
|
515 |
|
516 snd_pcm_sw_params_free (params); |
|
517 return 0; |
|
518 |
|
519 /* ERRORS */ |
|
520 no_config: |
|
521 { |
|
522 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
523 ("Unable to determine current swparams for playback: %s", |
|
524 snd_strerror (err))); |
|
525 snd_pcm_sw_params_free (params); |
|
526 return err; |
|
527 } |
|
528 start_threshold: |
|
529 { |
|
530 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
531 ("Unable to set start threshold mode for playback: %s", |
|
532 snd_strerror (err))); |
|
533 snd_pcm_sw_params_free (params); |
|
534 return err; |
|
535 } |
|
536 set_avail: |
|
537 { |
|
538 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
539 ("Unable to set avail min for playback: %s", snd_strerror (err))); |
|
540 snd_pcm_sw_params_free (params); |
|
541 return err; |
|
542 } |
|
543 #if !GST_CHECK_ALSA_VERSION(1,0,16) |
|
544 set_align: |
|
545 { |
|
546 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
547 ("Unable to set transfer align for playback: %s", snd_strerror (err))); |
|
548 snd_pcm_sw_params_free (params); |
|
549 return err; |
|
550 } |
|
551 #endif |
|
552 set_sw_params: |
|
553 { |
|
554 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
555 ("Unable to set sw params for playback: %s", snd_strerror (err))); |
|
556 snd_pcm_sw_params_free (params); |
|
557 return err; |
|
558 } |
|
559 } |
|
560 |
|
561 static gboolean |
|
562 alsasrc_parse_spec (GstAlsaSrc * alsa, GstRingBufferSpec * spec) |
|
563 { |
|
564 switch (spec->type) { |
|
565 case GST_BUFTYPE_LINEAR: |
|
566 alsa->format = snd_pcm_build_linear_format (spec->depth, spec->width, |
|
567 spec->sign ? 0 : 1, spec->bigend ? 1 : 0); |
|
568 break; |
|
569 case GST_BUFTYPE_FLOAT: |
|
570 switch (spec->format) { |
|
571 case GST_FLOAT32_LE: |
|
572 alsa->format = SND_PCM_FORMAT_FLOAT_LE; |
|
573 break; |
|
574 case GST_FLOAT32_BE: |
|
575 alsa->format = SND_PCM_FORMAT_FLOAT_BE; |
|
576 break; |
|
577 case GST_FLOAT64_LE: |
|
578 alsa->format = SND_PCM_FORMAT_FLOAT64_LE; |
|
579 break; |
|
580 case GST_FLOAT64_BE: |
|
581 alsa->format = SND_PCM_FORMAT_FLOAT64_BE; |
|
582 break; |
|
583 default: |
|
584 goto error; |
|
585 } |
|
586 break; |
|
587 case GST_BUFTYPE_A_LAW: |
|
588 alsa->format = SND_PCM_FORMAT_A_LAW; |
|
589 break; |
|
590 case GST_BUFTYPE_MU_LAW: |
|
591 alsa->format = SND_PCM_FORMAT_MU_LAW; |
|
592 break; |
|
593 default: |
|
594 goto error; |
|
595 |
|
596 } |
|
597 alsa->rate = spec->rate; |
|
598 alsa->channels = spec->channels; |
|
599 alsa->buffer_time = spec->buffer_time; |
|
600 alsa->period_time = spec->latency_time; |
|
601 alsa->access = SND_PCM_ACCESS_RW_INTERLEAVED; |
|
602 |
|
603 return TRUE; |
|
604 |
|
605 /* ERRORS */ |
|
606 error: |
|
607 { |
|
608 return FALSE; |
|
609 } |
|
610 } |
|
611 |
|
612 static gboolean |
|
613 gst_alsasrc_open (GstAudioSrc * asrc) |
|
614 { |
|
615 GstAlsaSrc *alsa; |
|
616 gint err; |
|
617 |
|
618 alsa = GST_ALSA_SRC (asrc); |
|
619 |
|
620 CHECK (snd_pcm_open (&alsa->handle, alsa->device, SND_PCM_STREAM_CAPTURE, |
|
621 SND_PCM_NONBLOCK), open_error); |
|
622 |
|
623 if (!alsa->mixer) |
|
624 alsa->mixer = gst_alsa_mixer_new (alsa->device, GST_ALSA_MIXER_CAPTURE); |
|
625 |
|
626 return TRUE; |
|
627 |
|
628 /* ERRORS */ |
|
629 open_error: |
|
630 { |
|
631 if (err == -EBUSY) { |
|
632 GST_ELEMENT_ERROR (alsa, RESOURCE, BUSY, |
|
633 (_("Could not open audio device for recording. " |
|
634 "Device is being used by another application.")), |
|
635 ("Device '%s' is busy", alsa->device)); |
|
636 } else { |
|
637 GST_ELEMENT_ERROR (alsa, RESOURCE, OPEN_READ, |
|
638 (_("Could not open audio device for recording.")), |
|
639 ("Recording open error on device '%s': %s", alsa->device, |
|
640 snd_strerror (err))); |
|
641 } |
|
642 return FALSE; |
|
643 } |
|
644 } |
|
645 |
|
646 static gboolean |
|
647 gst_alsasrc_prepare (GstAudioSrc * asrc, GstRingBufferSpec * spec) |
|
648 { |
|
649 GstAlsaSrc *alsa; |
|
650 gint err; |
|
651 |
|
652 alsa = GST_ALSA_SRC (asrc); |
|
653 |
|
654 if (!alsasrc_parse_spec (alsa, spec)) |
|
655 goto spec_parse; |
|
656 |
|
657 CHECK (snd_pcm_nonblock (alsa->handle, 0), non_block); |
|
658 |
|
659 CHECK (set_hwparams (alsa), hw_params_failed); |
|
660 CHECK (set_swparams (alsa), sw_params_failed); |
|
661 CHECK (snd_pcm_prepare (alsa->handle), prepare_failed); |
|
662 |
|
663 alsa->bytes_per_sample = spec->bytes_per_sample; |
|
664 spec->segsize = alsa->period_size * spec->bytes_per_sample; |
|
665 spec->segtotal = alsa->buffer_size / alsa->period_size; |
|
666 spec->silence_sample[0] = 0; |
|
667 spec->silence_sample[1] = 0; |
|
668 spec->silence_sample[2] = 0; |
|
669 spec->silence_sample[3] = 0; |
|
670 |
|
671 return TRUE; |
|
672 |
|
673 /* ERRORS */ |
|
674 spec_parse: |
|
675 { |
|
676 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
677 ("Error parsing spec")); |
|
678 return FALSE; |
|
679 } |
|
680 non_block: |
|
681 { |
|
682 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
683 ("Could not set device to blocking: %s", snd_strerror (err))); |
|
684 return FALSE; |
|
685 } |
|
686 hw_params_failed: |
|
687 { |
|
688 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
689 ("Setting of hwparams failed: %s", snd_strerror (err))); |
|
690 return FALSE; |
|
691 } |
|
692 sw_params_failed: |
|
693 { |
|
694 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
695 ("Setting of swparams failed: %s", snd_strerror (err))); |
|
696 return FALSE; |
|
697 } |
|
698 prepare_failed: |
|
699 { |
|
700 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
701 ("Prepare failed: %s", snd_strerror (err))); |
|
702 return FALSE; |
|
703 } |
|
704 } |
|
705 |
|
706 static gboolean |
|
707 gst_alsasrc_unprepare (GstAudioSrc * asrc) |
|
708 { |
|
709 GstAlsaSrc *alsa; |
|
710 gint err; |
|
711 |
|
712 alsa = GST_ALSA_SRC (asrc); |
|
713 |
|
714 CHECK (snd_pcm_drop (alsa->handle), drop); |
|
715 |
|
716 CHECK (snd_pcm_hw_free (alsa->handle), hw_free); |
|
717 |
|
718 CHECK (snd_pcm_nonblock (alsa->handle, 1), non_block); |
|
719 |
|
720 return TRUE; |
|
721 |
|
722 /* ERRORS */ |
|
723 drop: |
|
724 { |
|
725 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
726 ("Could not drop samples: %s", snd_strerror (err))); |
|
727 return FALSE; |
|
728 } |
|
729 hw_free: |
|
730 { |
|
731 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
732 ("Could not free hw params: %s", snd_strerror (err))); |
|
733 return FALSE; |
|
734 } |
|
735 non_block: |
|
736 { |
|
737 GST_ELEMENT_ERROR (alsa, RESOURCE, SETTINGS, (NULL), |
|
738 ("Could not set device to nonblocking: %s", snd_strerror (err))); |
|
739 return FALSE; |
|
740 } |
|
741 } |
|
742 |
|
743 static gboolean |
|
744 gst_alsasrc_close (GstAudioSrc * asrc) |
|
745 { |
|
746 GstAlsaSrc *alsa = GST_ALSA_SRC (asrc); |
|
747 |
|
748 snd_pcm_close (alsa->handle); |
|
749 |
|
750 if (alsa->mixer) { |
|
751 gst_alsa_mixer_free (alsa->mixer); |
|
752 alsa->mixer = NULL; |
|
753 } |
|
754 |
|
755 gst_caps_replace (&alsa->cached_caps, NULL); |
|
756 |
|
757 return TRUE; |
|
758 } |
|
759 |
|
760 /* |
|
761 * Underrun and suspend recovery |
|
762 */ |
|
763 static gint |
|
764 xrun_recovery (GstAlsaSrc * alsa, snd_pcm_t * handle, gint err) |
|
765 { |
|
766 GST_DEBUG_OBJECT (alsa, "xrun recovery %d", err); |
|
767 |
|
768 if (err == -EPIPE) { /* under-run */ |
|
769 err = snd_pcm_prepare (handle); |
|
770 if (err < 0) |
|
771 GST_WARNING_OBJECT (alsa, |
|
772 "Can't recovery from underrun, prepare failed: %s", |
|
773 snd_strerror (err)); |
|
774 return 0; |
|
775 } else if (err == -ESTRPIPE) { |
|
776 while ((err = snd_pcm_resume (handle)) == -EAGAIN) |
|
777 g_usleep (100); /* wait until the suspend flag is released */ |
|
778 |
|
779 if (err < 0) { |
|
780 err = snd_pcm_prepare (handle); |
|
781 if (err < 0) |
|
782 GST_WARNING_OBJECT (alsa, |
|
783 "Can't recovery from suspend, prepare failed: %s", |
|
784 snd_strerror (err)); |
|
785 } |
|
786 return 0; |
|
787 } |
|
788 return err; |
|
789 } |
|
790 |
|
791 static guint |
|
792 gst_alsasrc_read (GstAudioSrc * asrc, gpointer data, guint length) |
|
793 { |
|
794 GstAlsaSrc *alsa; |
|
795 gint err; |
|
796 gint cptr; |
|
797 gint16 *ptr; |
|
798 |
|
799 alsa = GST_ALSA_SRC (asrc); |
|
800 |
|
801 cptr = length / alsa->bytes_per_sample; |
|
802 ptr = data; |
|
803 |
|
804 GST_ALSA_SRC_LOCK (asrc); |
|
805 while (cptr > 0) { |
|
806 if ((err = snd_pcm_readi (alsa->handle, ptr, cptr)) < 0) { |
|
807 if (err == -EAGAIN) { |
|
808 GST_DEBUG_OBJECT (asrc, "Read error: %s", snd_strerror (err)); |
|
809 continue; |
|
810 } else if (xrun_recovery (alsa, alsa->handle, err) < 0) { |
|
811 goto read_error; |
|
812 } |
|
813 continue; |
|
814 } |
|
815 |
|
816 ptr += err * alsa->channels; |
|
817 cptr -= err; |
|
818 } |
|
819 GST_ALSA_SRC_UNLOCK (asrc); |
|
820 |
|
821 return length - cptr; |
|
822 |
|
823 read_error: |
|
824 { |
|
825 GST_ALSA_SRC_UNLOCK (asrc); |
|
826 return length; /* skip one period */ |
|
827 } |
|
828 } |
|
829 |
|
830 static guint |
|
831 gst_alsasrc_delay (GstAudioSrc * asrc) |
|
832 { |
|
833 GstAlsaSrc *alsa; |
|
834 snd_pcm_sframes_t delay; |
|
835 int res; |
|
836 |
|
837 alsa = GST_ALSA_SRC (asrc); |
|
838 |
|
839 res = snd_pcm_delay (alsa->handle, &delay); |
|
840 if (G_UNLIKELY (res < 0)) { |
|
841 GST_DEBUG_OBJECT (alsa, "snd_pcm_delay returned %d", res); |
|
842 delay = 0; |
|
843 } |
|
844 |
|
845 return CLAMP (delay, 0, alsa->buffer_size); |
|
846 } |
|
847 |
|
848 static void |
|
849 gst_alsasrc_reset (GstAudioSrc * asrc) |
|
850 { |
|
851 GstAlsaSrc *alsa; |
|
852 gint err; |
|
853 |
|
854 alsa = GST_ALSA_SRC (asrc); |
|
855 |
|
856 GST_ALSA_SRC_LOCK (asrc); |
|
857 GST_DEBUG_OBJECT (alsa, "drop"); |
|
858 CHECK (snd_pcm_drop (alsa->handle), drop_error); |
|
859 GST_DEBUG_OBJECT (alsa, "prepare"); |
|
860 CHECK (snd_pcm_prepare (alsa->handle), prepare_error); |
|
861 GST_DEBUG_OBJECT (alsa, "reset done"); |
|
862 GST_ALSA_SRC_UNLOCK (asrc); |
|
863 |
|
864 return; |
|
865 |
|
866 /* ERRORS */ |
|
867 drop_error: |
|
868 { |
|
869 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm drop error: %s", |
|
870 snd_strerror (err)); |
|
871 GST_ALSA_SRC_UNLOCK (asrc); |
|
872 return; |
|
873 } |
|
874 prepare_error: |
|
875 { |
|
876 GST_ERROR_OBJECT (alsa, "alsa-reset: pcm prepare error: %s", |
|
877 snd_strerror (err)); |
|
878 GST_ALSA_SRC_UNLOCK (asrc); |
|
879 return; |
|
880 } |
|
881 } |
|