annotate stream/audio_in.c @ 30674:553f53cffabf

Reset output color after each line.
author zuxy
date Tue, 23 Feb 2010 06:42:51 +0000
parents ce0122361a39
children b39155e98ac3
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
rev   line source
30426
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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1 /*
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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2 * This file is part of MPlayer.
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3 *
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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4 * MPlayer is free software; you can redistribute it and/or modify
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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5 * it under the terms of the GNU General Public License as published by
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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6 * the Free Software Foundation; either version 2 of the License, or
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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7 * (at your option) any later version.
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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8 *
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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9 * MPlayer is distributed in the hope that it will be useful,
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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10 * but WITHOUT ANY WARRANTY; without even the implied warranty of
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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11 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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12 * GNU General Public License for more details.
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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13 *
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14 * You should have received a copy of the GNU General Public License along
ce0122361a39 Add license header to all files missing it in the stream subdirectory.
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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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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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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 if (ai->alsa.device) 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 if (ai->oss.device) 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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c4434bdf6e51 tons of warning fixes, also some 10l bugfixes, including Dominik's PVA bug
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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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diff changeset
206 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret));
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diff changeset
207 if (ret == -EPIPE) {
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208 if (ai_alsa_xrun(ai) == 0) {
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diff changeset
209 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut);
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210 } else {
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diff changeset
211 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover);
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212 }
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diff changeset
213 }
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214 } else {
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diff changeset
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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diff changeset
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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diff changeset
226 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno));
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227 } else {
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diff changeset
228 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples);
7060
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229 }
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230 return -1;
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231 }
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232 return ret;
7843
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diff changeset
233 #endif
7060
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234 default:
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235 return -1;
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236 }
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237 }