Mercurial > mplayer.hg
annotate libmpdemux/audio_in.c @ 7898:4bb38736c381
10l... #ifdef doesn't work for enums :(
author | arpi |
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date | Thu, 24 Oct 2002 19:40:08 +0000 |
parents | ef4046febd24 |
children | 772d6d27fd66 |
rev | line source |
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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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audio in layer is used only bu tvi_v4l, don't compile for others.
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2 #include <stdlib.h> |
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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 #if defined(USE_TV) && defined(HAVE_TV_V4L) |
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8 |
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9 #include "audio_in.h" |
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10 #include "mp_msg.h" |
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11 #include <string.h> |
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12 #include <errno.h> |
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13 |
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14 // sanitizes ai structure before calling other functions |
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15 int audio_in_init(audio_in_t *ai, int type) |
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16 { |
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17 ai->type = type; |
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18 ai->setup = 0; |
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19 |
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20 ai->channels = -1; |
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21 ai->samplerate = -1; |
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22 ai->blocksize = -1; |
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23 ai->bytes_per_sample = -1; |
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24 ai->samplesize = -1; |
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25 |
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26 switch (ai->type) { |
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27 #ifdef HAVE_ALSA9 |
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28 case AUDIO_IN_ALSA: |
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29 ai->alsa.handle = NULL; |
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30 ai->alsa.log = NULL; |
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31 ai->alsa.device = strdup("default"); |
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32 return 0; |
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33 #endif |
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34 #ifdef USE_OSS_AUDIO |
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35 case AUDIO_IN_OSS: |
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36 ai->oss.audio_fd = -1; |
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37 ai->oss.device = strdup("/dev/dsp"); |
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38 return 0; |
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39 #endif |
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40 default: |
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41 return -1; |
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42 } |
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43 } |
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44 |
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45 int audio_in_setup(audio_in_t *ai) |
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46 { |
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47 |
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48 switch (ai->type) { |
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49 #ifdef HAVE_ALSA9 |
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50 case AUDIO_IN_ALSA: |
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51 if (ai_alsa_init(ai) < 0) return -1; |
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52 ai->setup = 1; |
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53 return 0; |
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54 #endif |
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55 #ifdef USE_OSS_AUDIO |
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56 case AUDIO_IN_OSS: |
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57 if (ai_oss_init(ai) < 0) return -1; |
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58 ai->setup = 1; |
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59 return 0; |
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60 #endif |
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61 default: |
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62 return -1; |
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63 } |
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64 } |
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65 |
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66 int audio_in_set_samplerate(audio_in_t *ai, int rate) |
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67 { |
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68 switch (ai->type) { |
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69 #ifdef HAVE_ALSA9 |
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70 case AUDIO_IN_ALSA: |
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71 ai->req_samplerate = rate; |
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72 if (!ai->setup) return 0; |
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73 if (ai_alsa_setup(ai) < 0) return -1; |
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74 return ai->samplerate; |
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75 #endif |
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76 #ifdef USE_OSS_AUDIO |
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77 case AUDIO_IN_OSS: |
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78 ai->req_samplerate = rate; |
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79 if (!ai->setup) return 0; |
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80 if (ai_oss_set_samplerate(ai) < 0) return -1; |
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81 return ai->samplerate; |
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82 #endif |
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83 default: |
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84 return -1; |
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85 } |
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86 } |
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87 |
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88 int audio_in_set_channels(audio_in_t *ai, int channels) |
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89 { |
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90 switch (ai->type) { |
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91 #ifdef HAVE_ALSA9 |
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92 case AUDIO_IN_ALSA: |
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93 ai->req_channels = channels; |
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94 if (!ai->setup) return 0; |
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95 if (ai_alsa_setup(ai) < 0) return -1; |
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96 return ai->channels; |
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97 #endif |
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98 #ifdef USE_OSS_AUDIO |
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99 case AUDIO_IN_OSS: |
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100 ai->req_channels = channels; |
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101 if (!ai->setup) return 0; |
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102 if (ai_oss_set_channels(ai) < 0) return -1; |
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103 return ai->channels; |
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104 #endif |
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105 default: |
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106 return -1; |
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107 } |
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108 } |
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109 |
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110 int audio_in_set_device(audio_in_t *ai, char *device) |
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111 { |
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112 int i; |
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113 if (ai->setup) return -1; |
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114 switch (ai->type) { |
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115 #ifdef HAVE_ALSA9 |
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116 case AUDIO_IN_ALSA: |
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117 if (ai->alsa.device) free(ai->alsa.device); |
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118 ai->alsa.device = strdup(device); |
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119 /* mplayer cannot handle colons in arguments */ |
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120 for (i = 0; i < (int)strlen(ai->alsa.device); i++) { |
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121 if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; |
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122 } |
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123 return 0; |
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124 #endif |
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125 #ifdef USE_OSS_AUDIO |
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126 case AUDIO_IN_OSS: |
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127 if (ai->oss.device) free(ai->oss.device); |
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128 ai->oss.device = strdup(device); |
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129 return 0; |
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130 #endif |
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131 default: |
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132 return -1; |
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133 } |
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134 } |
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135 |
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136 int audio_in_uninit(audio_in_t *ai) |
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137 { |
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138 if (ai->setup) { |
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139 switch (ai->type) { |
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140 #ifdef HAVE_ALSA9 |
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141 case AUDIO_IN_ALSA: |
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142 if (ai->alsa.log) |
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143 snd_output_close(ai->alsa.log); |
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144 if (ai->alsa.handle) { |
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145 snd_pcm_close(ai->alsa.handle); |
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146 } |
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147 ai->setup = 0; |
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148 return 0; |
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149 #endif |
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150 #ifdef USE_OSS_AUDIO |
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151 case AUDIO_IN_OSS: |
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152 close(ai->oss.audio_fd); |
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153 ai->setup = 0; |
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154 return 0; |
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155 #endif |
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156 } |
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157 } |
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158 return -1; |
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159 } |
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160 |
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161 int audio_in_start_capture(audio_in_t *ai) |
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162 { |
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163 switch (ai->type) { |
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164 #ifdef HAVE_ALSA9 |
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165 case AUDIO_IN_ALSA: |
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166 return snd_pcm_start(ai->alsa.handle); |
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167 #endif |
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168 #ifdef USE_OSS_AUDIO |
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169 case AUDIO_IN_OSS: |
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170 return 0; |
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171 #endif |
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172 default: |
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173 return -1; |
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174 } |
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175 } |
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176 |
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177 int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) |
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178 { |
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179 int ret; |
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180 |
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181 switch (ai->type) { |
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182 #ifdef HAVE_ALSA9 |
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183 case AUDIO_IN_ALSA: |
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184 ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); |
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185 if (ret != ai->alsa.chunk_size) { |
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186 if (ret < 0) { |
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187 mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", snd_strerror(ret)); |
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188 if (ret == -EPIPE) { |
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189 if (ai_alsa_xrun(ai) == 0) { |
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190 mp_msg(MSGT_TV, MSGL_ERR, "Recovered from cross-run, some frames may be left out!\n"); |
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191 } else { |
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192 mp_msg(MSGT_TV, MSGL_ERR, "Fatal error, cannot recover!\n"); |
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193 } |
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194 } |
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195 } else { |
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196 mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); |
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197 } |
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198 return -1; |
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199 } |
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200 return ret; |
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201 #endif |
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202 #ifdef USE_OSS_AUDIO |
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203 case AUDIO_IN_OSS: |
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204 ret = read(ai->oss.audio_fd, buffer, ai->blocksize); |
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205 if (ret != ai->blocksize) { |
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206 if (ret < 0) { |
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207 mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", strerror(errno)); |
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208 } else { |
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209 mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); |
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210 } |
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211 return -1; |
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212 } |
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213 return ret; |
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214 #endif |
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215 default: |
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216 return -1; |
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217 } |
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218 } |
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219 |
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220 #endif |