Mercurial > mplayer.hg
annotate stream/audio_in.c @ 36579:c11abb032606
mplayer: Simplify initial playtree/file code.
Also avoids a theoretical NULL dereference if play_tree_iter_step
does not return an entry.
"Theoretical" since I don't know if or how this case would
be possible to trigger, so it might in fact be dead code.
author | reimar |
---|---|
date | Mon, 20 Jan 2014 22:08:45 +0000 |
parents | b39155e98ac3 |
children |
rev | line source |
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1 /* |
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2 * This file is part of MPlayer. |
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3 * |
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4 * MPlayer is free software; you can redistribute it and/or modify |
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5 * it under the terms of the GNU General Public License as published by |
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6 * the Free Software Foundation; either version 2 of the License, or |
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7 * (at your option) any later version. |
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8 * |
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9 * MPlayer is distributed in the hope that it will be useful, |
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10 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
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11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the |
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12 * GNU General Public License for more details. |
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13 * |
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14 * You should have received a copy of the GNU General Public License along |
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15 * with MPlayer; if not, write to the Free Software Foundation, Inc., |
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16 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. |
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17 */ |
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18 |
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19 #include <stdio.h> |
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20 #include <stdlib.h> |
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tons of warning fixes, also some 10l bugfixes, including Dominik's PVA bug
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21 #include <unistd.h> |
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22 |
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23 #include "config.h" |
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24 |
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25 #include "audio_in.h" |
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26 #include "mp_msg.h" |
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libmpdemux translatables to help_mp part 1 / mp_msg calls / try 2
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27 #include "help_mp.h" |
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28 #include <string.h> |
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29 #include <errno.h> |
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30 |
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31 // sanitizes ai structure before calling other functions |
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32 int audio_in_init(audio_in_t *ai, int type) |
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33 { |
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34 ai->type = type; |
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35 ai->setup = 0; |
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36 |
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37 ai->channels = -1; |
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38 ai->samplerate = -1; |
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39 ai->blocksize = -1; |
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40 ai->bytes_per_sample = -1; |
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41 ai->samplesize = -1; |
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42 |
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43 switch (ai->type) { |
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44 #ifdef CONFIG_ALSA |
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45 case AUDIO_IN_ALSA: |
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46 ai->alsa.handle = NULL; |
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47 ai->alsa.log = NULL; |
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48 ai->alsa.device = strdup("default"); |
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49 return 0; |
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50 #endif |
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51 #ifdef CONFIG_OSS_AUDIO |
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52 case AUDIO_IN_OSS: |
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53 ai->oss.audio_fd = -1; |
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54 ai->oss.device = strdup("/dev/dsp"); |
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55 return 0; |
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56 #endif |
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57 default: |
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58 return -1; |
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59 } |
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60 } |
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61 |
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62 int audio_in_setup(audio_in_t *ai) |
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63 { |
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64 |
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65 switch (ai->type) { |
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66 #ifdef CONFIG_ALSA |
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67 case AUDIO_IN_ALSA: |
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68 if (ai_alsa_init(ai) < 0) return -1; |
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69 ai->setup = 1; |
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70 return 0; |
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71 #endif |
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72 #ifdef CONFIG_OSS_AUDIO |
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73 case AUDIO_IN_OSS: |
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74 if (ai_oss_init(ai) < 0) return -1; |
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75 ai->setup = 1; |
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76 return 0; |
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77 #endif |
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78 default: |
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79 return -1; |
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80 } |
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81 } |
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82 |
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83 int audio_in_set_samplerate(audio_in_t *ai, int rate) |
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84 { |
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85 switch (ai->type) { |
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86 #ifdef CONFIG_ALSA |
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87 case AUDIO_IN_ALSA: |
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88 ai->req_samplerate = rate; |
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89 if (!ai->setup) return 0; |
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90 if (ai_alsa_setup(ai) < 0) return -1; |
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91 return ai->samplerate; |
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92 #endif |
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93 #ifdef CONFIG_OSS_AUDIO |
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94 case AUDIO_IN_OSS: |
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95 ai->req_samplerate = rate; |
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96 if (!ai->setup) return 0; |
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97 if (ai_oss_set_samplerate(ai) < 0) return -1; |
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98 return ai->samplerate; |
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99 #endif |
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100 default: |
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101 return -1; |
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102 } |
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103 } |
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104 |
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105 int audio_in_set_channels(audio_in_t *ai, int channels) |
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106 { |
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107 switch (ai->type) { |
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108 #ifdef CONFIG_ALSA |
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109 case AUDIO_IN_ALSA: |
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110 ai->req_channels = channels; |
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111 if (!ai->setup) return 0; |
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112 if (ai_alsa_setup(ai) < 0) return -1; |
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113 return ai->channels; |
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114 #endif |
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115 #ifdef CONFIG_OSS_AUDIO |
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116 case AUDIO_IN_OSS: |
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117 ai->req_channels = channels; |
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118 if (!ai->setup) return 0; |
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119 if (ai_oss_set_channels(ai) < 0) return -1; |
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120 return ai->channels; |
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121 #endif |
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122 default: |
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123 return -1; |
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124 } |
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125 } |
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126 |
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127 int audio_in_set_device(audio_in_t *ai, char *device) |
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128 { |
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129 #ifdef CONFIG_ALSA |
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130 int i; |
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131 #endif |
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132 if (ai->setup) return -1; |
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133 switch (ai->type) { |
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134 #ifdef CONFIG_ALSA |
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135 case AUDIO_IN_ALSA: |
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136 free(ai->alsa.device); |
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137 ai->alsa.device = strdup(device); |
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138 /* mplayer cannot handle colons in arguments */ |
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139 for (i = 0; i < (int)strlen(ai->alsa.device); i++) { |
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140 if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; |
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141 } |
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142 return 0; |
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143 #endif |
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144 #ifdef CONFIG_OSS_AUDIO |
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145 case AUDIO_IN_OSS: |
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146 free(ai->oss.device); |
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147 ai->oss.device = strdup(device); |
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148 return 0; |
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149 #endif |
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150 default: |
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151 return -1; |
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152 } |
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153 } |
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154 |
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155 int audio_in_uninit(audio_in_t *ai) |
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156 { |
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157 if (ai->setup) { |
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158 switch (ai->type) { |
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159 #ifdef CONFIG_ALSA |
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160 case AUDIO_IN_ALSA: |
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161 if (ai->alsa.log) |
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162 snd_output_close(ai->alsa.log); |
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163 if (ai->alsa.handle) { |
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164 snd_pcm_close(ai->alsa.handle); |
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165 } |
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166 ai->setup = 0; |
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167 return 0; |
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168 #endif |
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169 #ifdef CONFIG_OSS_AUDIO |
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170 case AUDIO_IN_OSS: |
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171 close(ai->oss.audio_fd); |
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172 ai->setup = 0; |
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173 return 0; |
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174 #endif |
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175 } |
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176 } |
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177 return -1; |
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178 } |
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179 |
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180 int audio_in_start_capture(audio_in_t *ai) |
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181 { |
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182 switch (ai->type) { |
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183 #ifdef CONFIG_ALSA |
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184 case AUDIO_IN_ALSA: |
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185 return snd_pcm_start(ai->alsa.handle); |
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186 #endif |
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187 #ifdef CONFIG_OSS_AUDIO |
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188 case AUDIO_IN_OSS: |
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189 return 0; |
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190 #endif |
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191 default: |
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192 return -1; |
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193 } |
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194 } |
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195 |
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196 int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) |
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197 { |
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198 int ret; |
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199 |
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200 switch (ai->type) { |
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201 #ifdef CONFIG_ALSA |
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202 case AUDIO_IN_ALSA: |
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203 ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); |
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204 if (ret != ai->alsa.chunk_size) { |
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205 if (ret < 0) { |
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206 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret)); |
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207 if (ret == -EPIPE) { |
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208 if (ai_alsa_xrun(ai) == 0) { |
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209 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut); |
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210 } else { |
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211 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover); |
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212 } |
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213 } |
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214 } else { |
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215 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); |
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216 } |
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217 return -1; |
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218 } |
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219 return ret; |
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220 #endif |
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221 #ifdef CONFIG_OSS_AUDIO |
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222 case AUDIO_IN_OSS: |
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223 ret = read(ai->oss.audio_fd, buffer, ai->blocksize); |
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224 if (ret != ai->blocksize) { |
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225 if (ret < 0) { |
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226 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno)); |
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227 } else { |
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228 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); |
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229 } |
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230 return -1; |
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231 } |
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232 return ret; |
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233 #endif |
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234 default: |
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235 return -1; |
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236 } |
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237 } |