Mercurial > mplayer.hg
annotate libmpdemux/audio_in.c @ 15918:8f6c98310f38
initial translation, synced with 1.12
author | gabrov |
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date | Mon, 04 Jul 2005 18:29:48 +0000 |
parents | 66e491c35dc8 |
children | dfbe8cd0e081 |
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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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) { |
11775 | 27 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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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) { |
11775 | 49 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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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) { |
11775 | 69 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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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) { |
11775 | 91 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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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 { |
11775 | 112 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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113 int i; |
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114 #endif |
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115 if (ai->setup) return -1; |
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116 switch (ai->type) { |
11775 | 117 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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118 case AUDIO_IN_ALSA: |
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119 if (ai->alsa.device) free(ai->alsa.device); |
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120 ai->alsa.device = strdup(device); |
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121 /* mplayer cannot handle colons in arguments */ |
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122 for (i = 0; i < (int)strlen(ai->alsa.device); i++) { |
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123 if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; |
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124 } |
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125 return 0; |
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126 #endif |
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127 #ifdef USE_OSS_AUDIO |
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128 case AUDIO_IN_OSS: |
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129 if (ai->oss.device) free(ai->oss.device); |
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130 ai->oss.device = strdup(device); |
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131 return 0; |
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132 #endif |
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133 default: |
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134 return -1; |
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135 } |
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136 } |
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137 |
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138 int audio_in_uninit(audio_in_t *ai) |
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139 { |
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140 if (ai->setup) { |
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141 switch (ai->type) { |
11775 | 142 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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143 case AUDIO_IN_ALSA: |
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144 if (ai->alsa.log) |
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145 snd_output_close(ai->alsa.log); |
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146 if (ai->alsa.handle) { |
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147 snd_pcm_close(ai->alsa.handle); |
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148 } |
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149 ai->setup = 0; |
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150 return 0; |
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151 #endif |
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152 #ifdef USE_OSS_AUDIO |
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153 case AUDIO_IN_OSS: |
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154 close(ai->oss.audio_fd); |
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155 ai->setup = 0; |
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156 return 0; |
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157 #endif |
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158 } |
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159 } |
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160 return -1; |
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161 } |
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162 |
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163 int audio_in_start_capture(audio_in_t *ai) |
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164 { |
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165 switch (ai->type) { |
11775 | 166 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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167 case AUDIO_IN_ALSA: |
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168 return snd_pcm_start(ai->alsa.handle); |
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169 #endif |
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170 #ifdef USE_OSS_AUDIO |
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171 case AUDIO_IN_OSS: |
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172 return 0; |
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173 #endif |
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174 default: |
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175 return -1; |
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176 } |
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177 } |
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178 |
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179 int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) |
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180 { |
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181 int ret; |
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182 |
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183 switch (ai->type) { |
11775 | 184 #if defined(HAVE_ALSA9) || defined(HAVE_ALSA1X) |
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185 case AUDIO_IN_ALSA: |
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186 ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); |
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187 if (ret != ai->alsa.chunk_size) { |
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188 if (ret < 0) { |
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189 mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", snd_strerror(ret)); |
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190 if (ret == -EPIPE) { |
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191 if (ai_alsa_xrun(ai) == 0) { |
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192 mp_msg(MSGT_TV, MSGL_ERR, "Recovered from cross-run, some frames may be left out!\n"); |
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193 } else { |
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194 mp_msg(MSGT_TV, MSGL_ERR, "Fatal error, cannot recover!\n"); |
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195 } |
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196 } |
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197 } else { |
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198 mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); |
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199 } |
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200 return -1; |
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201 } |
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202 return ret; |
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203 #endif |
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204 #ifdef USE_OSS_AUDIO |
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205 case AUDIO_IN_OSS: |
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206 ret = read(ai->oss.audio_fd, buffer, ai->blocksize); |
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207 if (ret != ai->blocksize) { |
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208 if (ret < 0) { |
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209 mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", strerror(errno)); |
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210 } else { |
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211 mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); |
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212 } |
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213 return -1; |
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214 } |
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215 return ret; |
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216 #endif |
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217 default: |
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218 return -1; |
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219 } |
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220 } |
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221 |
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222 #endif |