annotate stream/audio_in.c @ 25308:7fdb9b2b6770

Fix printf format string length modifiers, removes the warnings: asfinfo.c: In function 'print_wave_header': asfinfo.c:114: warning: format '%d' expects type 'int', but argument 2 has type 'long int' asfinfo.c:115: warning: format '%d' expects type 'int', but argument 2 has type 'long int' asfinfo.c: In function 'print_video_header': asfinfo.c:140: warning: format '%d' expects type 'int', but argument 2 has type 'long int' asfinfo.c:141: warning: format '%d' expects type 'int', but argument 2 has type 'long int' asfinfo.c:142: warning: format '%d' expects type 'int', but argument 2 has type 'long int' asfinfo.c:145: warning: format '%d' expects type 'int', but argument 2 has type 'long int' asfinfo.c:146: warning: format '%d' expects type 'int', but argument 2 has type 'long int' asfinfo.c: In function 'main': asfinfo.c:174: warning: format '%X' expects type 'unsigned int', but argument 2 has type 'long int' asfinfo.c:220: warning: format '%d' expects type 'int', but argument 6 has type 'long unsigned int'
author diego
date Mon, 10 Dec 2007 13:11:26 +0000
parents 64d82a45a05d
children e7c989f7a7c9
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
rev   line source
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13fcab6fde41 audio in layer is used only bu tvi_v4l, don't compile for others.
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1 #include <stdio.h>
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2 #include <stdlib.h>
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c4434bdf6e51 tons of warning fixes, also some 10l bugfixes, including Dominik's PVA bug
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3 #include <unistd.h>
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4
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5 #include "config.h"
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6
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7 #include "audio_in.h"
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8 #include "mp_msg.h"
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9 #include "help_mp.h"
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10 #include <string.h>
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11 #include <errno.h>
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12
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13 // sanitizes ai structure before calling other functions
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14 int audio_in_init(audio_in_t *ai, int type)
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15 {
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16 ai->type = type;
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17 ai->setup = 0;
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18
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19 ai->channels = -1;
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20 ai->samplerate = -1;
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21 ai->blocksize = -1;
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22 ai->bytes_per_sample = -1;
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23 ai->samplesize = -1;
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24
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25 switch (ai->type) {
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26 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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27 case AUDIO_IN_ALSA:
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28 ai->alsa.handle = NULL;
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29 ai->alsa.log = NULL;
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30 ai->alsa.device = strdup("default");
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31 return 0;
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32 #endif
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33 #ifdef USE_OSS_AUDIO
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34 case AUDIO_IN_OSS:
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35 ai->oss.audio_fd = -1;
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36 ai->oss.device = strdup("/dev/dsp");
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37 return 0;
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38 #endif
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39 default:
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40 return -1;
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41 }
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42 }
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43
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44 int audio_in_setup(audio_in_t *ai)
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45 {
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46
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47 switch (ai->type) {
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48 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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49 case AUDIO_IN_ALSA:
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50 if (ai_alsa_init(ai) < 0) return -1;
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51 ai->setup = 1;
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52 return 0;
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53 #endif
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54 #ifdef USE_OSS_AUDIO
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55 case AUDIO_IN_OSS:
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56 if (ai_oss_init(ai) < 0) return -1;
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57 ai->setup = 1;
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58 return 0;
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59 #endif
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60 default:
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61 return -1;
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62 }
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63 }
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64
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65 int audio_in_set_samplerate(audio_in_t *ai, int rate)
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66 {
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67 switch (ai->type) {
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68 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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69 case AUDIO_IN_ALSA:
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70 ai->req_samplerate = rate;
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71 if (!ai->setup) return 0;
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72 if (ai_alsa_setup(ai) < 0) return -1;
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73 return ai->samplerate;
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74 #endif
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75 #ifdef USE_OSS_AUDIO
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76 case AUDIO_IN_OSS:
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77 ai->req_samplerate = rate;
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78 if (!ai->setup) return 0;
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79 if (ai_oss_set_samplerate(ai) < 0) return -1;
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80 return ai->samplerate;
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81 #endif
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82 default:
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83 return -1;
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84 }
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85 }
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86
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87 int audio_in_set_channels(audio_in_t *ai, int channels)
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88 {
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89 switch (ai->type) {
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90 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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91 case AUDIO_IN_ALSA:
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92 ai->req_channels = channels;
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93 if (!ai->setup) return 0;
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94 if (ai_alsa_setup(ai) < 0) return -1;
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95 return ai->channels;
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96 #endif
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97 #ifdef USE_OSS_AUDIO
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98 case AUDIO_IN_OSS:
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99 ai->req_channels = channels;
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100 if (!ai->setup) return 0;
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101 if (ai_oss_set_channels(ai) < 0) return -1;
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102 return ai->channels;
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103 #endif
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104 default:
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105 return -1;
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106 }
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107 }
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108
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109 int audio_in_set_device(audio_in_t *ai, char *device)
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110 {
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111 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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112 int i;
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113 #endif
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114 if (ai->setup) return -1;
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115 switch (ai->type) {
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116 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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117 case AUDIO_IN_ALSA:
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118 if (ai->alsa.device) free(ai->alsa.device);
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119 ai->alsa.device = strdup(device);
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120 /* mplayer cannot handle colons in arguments */
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121 for (i = 0; i < (int)strlen(ai->alsa.device); i++) {
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122 if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':';
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123 }
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124 return 0;
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125 #endif
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126 #ifdef USE_OSS_AUDIO
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127 case AUDIO_IN_OSS:
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128 if (ai->oss.device) free(ai->oss.device);
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129 ai->oss.device = strdup(device);
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130 return 0;
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131 #endif
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132 default:
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133 return -1;
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134 }
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135 }
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136
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137 int audio_in_uninit(audio_in_t *ai)
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138 {
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139 if (ai->setup) {
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140 switch (ai->type) {
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141 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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142 case AUDIO_IN_ALSA:
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143 if (ai->alsa.log)
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144 snd_output_close(ai->alsa.log);
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145 if (ai->alsa.handle) {
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146 snd_pcm_close(ai->alsa.handle);
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147 }
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148 ai->setup = 0;
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149 return 0;
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150 #endif
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151 #ifdef USE_OSS_AUDIO
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152 case AUDIO_IN_OSS:
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153 close(ai->oss.audio_fd);
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154 ai->setup = 0;
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155 return 0;
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156 #endif
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157 }
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158 }
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159 return -1;
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160 }
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161
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162 int audio_in_start_capture(audio_in_t *ai)
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163 {
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164 switch (ai->type) {
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165 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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166 case AUDIO_IN_ALSA:
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167 return snd_pcm_start(ai->alsa.handle);
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168 #endif
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169 #ifdef USE_OSS_AUDIO
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170 case AUDIO_IN_OSS:
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171 return 0;
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172 #endif
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173 default:
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174 return -1;
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175 }
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176 }
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177
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178 int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer)
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179 {
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180 int ret;
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181
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182 switch (ai->type) {
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183 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X)
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184 case AUDIO_IN_ALSA:
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185 ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size);
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186 if (ret != ai->alsa.chunk_size) {
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187 if (ret < 0) {
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188 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret));
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189 if (ret == -EPIPE) {
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190 if (ai_alsa_xrun(ai) == 0) {
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191 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut);
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192 } else {
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193 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover);
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194 }
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195 }
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196 } else {
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197 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples);
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198 }
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199 return -1;
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200 }
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201 return ret;
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202 #endif
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203 #ifdef USE_OSS_AUDIO
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204 case AUDIO_IN_OSS:
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205 ret = read(ai->oss.audio_fd, buffer, ai->blocksize);
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206 if (ret != ai->blocksize) {
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207 if (ret < 0) {
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diff changeset
208 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno));
7060
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209 } else {
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diff changeset
210 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples);
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211 }
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212 return -1;
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213 }
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214 return ret;
7843
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215 #endif
7060
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216 default:
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217 return -1;
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218 }
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219 }