Mercurial > mplayer.hg
annotate libmpdemux/audio_in.c @ 17094:cb35587c8156
fix
vf_phase.c:101: warning: suggest parentheses around + or - inside shift
vf_phase.c:102: warning: suggest parentheses around + or - inside shift
vf_phase.c:105: warning: suggest parentheses around + or - inside shift
vf_phase.c:106: warning: suggest parentheses around + or - inside shift
vf_phase.c:112: warning: suggest parentheses around + or - inside shift
vf_phase.c:113: warning: suggest parentheses around + or - inside shift
vf_phase.c:116: warning: suggest parentheses around + or - inside shift
vf_phase.c:117: warning: suggest parentheses around + or - inside shift
vf_phase.c:123: warning: suggest parentheses around + or - inside shift
vf_phase.c:124: warning: suggest parentheses around + or - inside shift
vf_phase.c:127: warning: suggest parentheses around + or - inside shift
vf_phase.c:128: warning: suggest parentheses around + or - inside shift
vf_phase.c:134: warning: suggest parentheses around + or - inside shift
vf_phase.c:135: warning: suggest parentheses around + or - inside shift
vf_phase.c:136: warning: suggest parentheses around + or - inside shift
vf_phase.c:139: warning: suggest parentheses around + or - inside shift
vf_phase.c:140: warning: suggest parentheses around + or - inside shift
vf_phase.c:141: warning: suggest parentheses around + or - inside shift
author | rathann |
---|---|
date | Mon, 05 Dec 2005 01:28:59 +0000 |
parents | dfbe8cd0e081 |
children | d2d9d011203f |
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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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 |
10537 | 7 #if defined(USE_TV) && (defined(HAVE_TV_V4L) || defined(HAVE_TV_V4L2)) |
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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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libmpdemux translatables to help_mp part 1 / mp_msg calls / try 2
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11 #include "help_mp.h" |
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12 #include <string.h> |
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13 #include <errno.h> |
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14 |
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15 // sanitizes ai structure before calling other functions |
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16 int audio_in_init(audio_in_t *ai, int type) |
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17 { |
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18 ai->type = type; |
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19 ai->setup = 0; |
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20 |
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21 ai->channels = -1; |
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22 ai->samplerate = -1; |
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23 ai->blocksize = -1; |
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24 ai->bytes_per_sample = -1; |
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25 ai->samplesize = -1; |
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26 |
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27 switch (ai->type) { |
11775 | 28 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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29 case AUDIO_IN_ALSA: |
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30 ai->alsa.handle = NULL; |
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31 ai->alsa.log = NULL; |
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32 ai->alsa.device = strdup("default"); |
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33 return 0; |
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34 #endif |
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35 #ifdef USE_OSS_AUDIO |
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36 case AUDIO_IN_OSS: |
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37 ai->oss.audio_fd = -1; |
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38 ai->oss.device = strdup("/dev/dsp"); |
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39 return 0; |
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40 #endif |
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41 default: |
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42 return -1; |
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43 } |
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44 } |
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45 |
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46 int audio_in_setup(audio_in_t *ai) |
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47 { |
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48 |
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49 switch (ai->type) { |
11775 | 50 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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51 case AUDIO_IN_ALSA: |
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52 if (ai_alsa_init(ai) < 0) return -1; |
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53 ai->setup = 1; |
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54 return 0; |
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55 #endif |
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56 #ifdef USE_OSS_AUDIO |
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57 case AUDIO_IN_OSS: |
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58 if (ai_oss_init(ai) < 0) return -1; |
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59 ai->setup = 1; |
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60 return 0; |
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61 #endif |
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62 default: |
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63 return -1; |
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64 } |
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65 } |
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66 |
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67 int audio_in_set_samplerate(audio_in_t *ai, int rate) |
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68 { |
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69 switch (ai->type) { |
11775 | 70 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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71 case AUDIO_IN_ALSA: |
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72 ai->req_samplerate = rate; |
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73 if (!ai->setup) return 0; |
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74 if (ai_alsa_setup(ai) < 0) return -1; |
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75 return ai->samplerate; |
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76 #endif |
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77 #ifdef USE_OSS_AUDIO |
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78 case AUDIO_IN_OSS: |
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79 ai->req_samplerate = rate; |
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80 if (!ai->setup) return 0; |
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81 if (ai_oss_set_samplerate(ai) < 0) return -1; |
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82 return ai->samplerate; |
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83 #endif |
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84 default: |
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85 return -1; |
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86 } |
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87 } |
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88 |
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89 int audio_in_set_channels(audio_in_t *ai, int channels) |
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90 { |
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91 switch (ai->type) { |
11775 | 92 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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93 case AUDIO_IN_ALSA: |
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94 ai->req_channels = channels; |
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95 if (!ai->setup) return 0; |
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96 if (ai_alsa_setup(ai) < 0) return -1; |
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97 return ai->channels; |
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98 #endif |
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99 #ifdef USE_OSS_AUDIO |
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100 case AUDIO_IN_OSS: |
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101 ai->req_channels = channels; |
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102 if (!ai->setup) return 0; |
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103 if (ai_oss_set_channels(ai) < 0) return -1; |
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104 return ai->channels; |
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105 #endif |
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106 default: |
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107 return -1; |
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108 } |
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109 } |
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110 |
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111 int audio_in_set_device(audio_in_t *ai, char *device) |
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112 { |
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114 int i; |
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115 #endif |
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116 if (ai->setup) return -1; |
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117 switch (ai->type) { |
11775 | 118 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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119 case AUDIO_IN_ALSA: |
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120 if (ai->alsa.device) free(ai->alsa.device); |
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121 ai->alsa.device = strdup(device); |
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122 /* mplayer cannot handle colons in arguments */ |
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123 for (i = 0; i < (int)strlen(ai->alsa.device); i++) { |
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124 if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; |
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125 } |
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126 return 0; |
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127 #endif |
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128 #ifdef USE_OSS_AUDIO |
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129 case AUDIO_IN_OSS: |
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130 if (ai->oss.device) free(ai->oss.device); |
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131 ai->oss.device = strdup(device); |
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132 return 0; |
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133 #endif |
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134 default: |
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135 return -1; |
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136 } |
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137 } |
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138 |
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139 int audio_in_uninit(audio_in_t *ai) |
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140 { |
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141 if (ai->setup) { |
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142 switch (ai->type) { |
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144 case AUDIO_IN_ALSA: |
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145 if (ai->alsa.log) |
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146 snd_output_close(ai->alsa.log); |
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147 if (ai->alsa.handle) { |
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148 snd_pcm_close(ai->alsa.handle); |
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149 } |
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150 ai->setup = 0; |
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151 return 0; |
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152 #endif |
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153 #ifdef USE_OSS_AUDIO |
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154 case AUDIO_IN_OSS: |
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155 close(ai->oss.audio_fd); |
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156 ai->setup = 0; |
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157 return 0; |
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158 #endif |
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159 } |
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160 } |
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161 return -1; |
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162 } |
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163 |
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164 int audio_in_start_capture(audio_in_t *ai) |
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165 { |
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166 switch (ai->type) { |
11775 | 167 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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168 case AUDIO_IN_ALSA: |
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169 return snd_pcm_start(ai->alsa.handle); |
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170 #endif |
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171 #ifdef USE_OSS_AUDIO |
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172 case AUDIO_IN_OSS: |
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173 return 0; |
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174 #endif |
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175 default: |
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176 return -1; |
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177 } |
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178 } |
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179 |
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180 int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) |
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181 { |
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182 int ret; |
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183 |
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184 switch (ai->type) { |
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186 case AUDIO_IN_ALSA: |
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187 ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); |
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188 if (ret != ai->alsa.chunk_size) { |
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189 if (ret < 0) { |
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190 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, snd_strerror(ret)); |
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191 if (ret == -EPIPE) { |
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192 if (ai_alsa_xrun(ai) == 0) { |
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193 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_XRUNSomeFramesMayBeLeftOut); |
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194 } else { |
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195 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrFatalCannotRecover); |
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196 } |
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197 } |
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198 } else { |
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199 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); |
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200 } |
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201 return -1; |
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202 } |
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203 return ret; |
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204 #endif |
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205 #ifdef USE_OSS_AUDIO |
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206 case AUDIO_IN_OSS: |
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207 ret = read(ai->oss.audio_fd, buffer, ai->blocksize); |
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208 if (ret != ai->blocksize) { |
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209 if (ret < 0) { |
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210 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno)); |
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211 } else { |
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212 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_NotEnoughSamples); |
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213 } |
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214 return -1; |
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215 } |
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216 return ret; |
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217 #endif |
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218 default: |
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219 return -1; |
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220 } |
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221 } |
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audio in layer is used only bu tvi_v4l, don't compile for others.
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222 |
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223 #endif |