Mercurial > mplayer.hg
annotate libmpdemux/audio_in.c @ 7217:7527dc997596
rewrite eq interface and add *.rm to fs
author | pontscho |
---|---|
date | Sun, 01 Sep 2002 11:28:18 +0000 |
parents | 13fcab6fde41 |
children | c4434bdf6e51 |
rev | line source |
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7213
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1 #include <stdio.h> |
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2 #include <stdlib.h> |
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3 |
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4 #include "config.h" |
7213
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5 |
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6 #if defined(USE_TV) && defined(HAVE_TV_V4L) |
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7 |
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8 #include "audio_in.h" |
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9 #include "mp_msg.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 #ifdef HAVE_ALSA9 |
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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 case AUDIO_IN_OSS: |
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34 ai->oss.audio_fd = -1; |
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35 ai->oss.device = strdup("/dev/dsp"); |
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36 return 0; |
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37 default: |
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38 return -1; |
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39 } |
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40 } |
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41 |
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42 int audio_in_setup(audio_in_t *ai) |
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43 { |
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44 int err; |
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45 |
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46 switch (ai->type) { |
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47 #ifdef HAVE_ALSA9 |
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48 case AUDIO_IN_ALSA: |
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49 if (ai_alsa_init(ai) < 0) return -1; |
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50 ai->setup = 1; |
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51 return 0; |
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52 #endif |
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53 case AUDIO_IN_OSS: |
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54 if (ai_oss_init(ai) < 0) return -1; |
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55 ai->setup = 1; |
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56 return 0; |
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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_set_samplerate(audio_in_t *ai, int rate) |
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63 { |
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64 switch (ai->type) { |
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65 #ifdef HAVE_ALSA9 |
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66 case AUDIO_IN_ALSA: |
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67 ai->req_samplerate = rate; |
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68 if (!ai->setup) return 0; |
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69 if (ai_alsa_setup(ai) < 0) return -1; |
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70 return ai->samplerate; |
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71 #endif |
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72 case AUDIO_IN_OSS: |
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73 ai->req_samplerate = rate; |
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74 if (!ai->setup) return 0; |
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75 if (ai_oss_set_samplerate(ai) < 0) return -1; |
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76 return ai->samplerate; |
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77 default: |
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78 return -1; |
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79 } |
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80 } |
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81 |
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82 int audio_in_set_channels(audio_in_t *ai, int channels) |
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83 { |
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84 switch (ai->type) { |
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85 #ifdef HAVE_ALSA9 |
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86 case AUDIO_IN_ALSA: |
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87 ai->req_channels = channels; |
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88 if (!ai->setup) return 0; |
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89 if (ai_alsa_setup(ai) < 0) return -1; |
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90 return ai->channels; |
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91 #endif |
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92 case AUDIO_IN_OSS: |
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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_oss_set_channels(ai) < 0) return -1; |
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96 return ai->channels; |
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97 default: |
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98 return -1; |
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99 } |
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100 } |
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101 |
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102 int audio_in_set_device(audio_in_t *ai, char *device) |
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103 { |
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104 int i; |
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105 if (ai->setup) return -1; |
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106 switch (ai->type) { |
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107 #ifdef HAVE_ALSA9 |
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108 case AUDIO_IN_ALSA: |
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109 if (ai->alsa.device) free(ai->alsa.device); |
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110 ai->alsa.device = strdup(device); |
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111 /* mplayer cannot handle colons in arguments */ |
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112 for (i = 0; i < strlen(ai->alsa.device); i++) { |
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113 if (ai->alsa.device[i] == '.') ai->alsa.device[i] = ':'; |
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114 } |
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115 return 0; |
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116 #endif |
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117 case AUDIO_IN_OSS: |
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118 if (ai->oss.device) free(ai->oss.device); |
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119 ai->oss.device = strdup(device); |
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120 return 0; |
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121 default: |
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122 return -1; |
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123 } |
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124 } |
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125 |
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126 int audio_in_uninit(audio_in_t *ai) |
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127 { |
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128 if (ai->setup) { |
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129 switch (ai->type) { |
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130 #ifdef HAVE_ALSA9 |
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131 case AUDIO_IN_ALSA: |
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132 if (ai->alsa.log) |
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133 snd_output_close(ai->alsa.log); |
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134 if (ai->alsa.handle) { |
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135 snd_pcm_close(ai->alsa.handle); |
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136 } |
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137 ai->setup = 0; |
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138 return 0; |
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139 #endif |
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140 case AUDIO_IN_OSS: |
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141 close(ai->oss.audio_fd); |
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142 ai->setup = 0; |
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143 return 0; |
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144 default: |
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145 return -1; |
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146 } |
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147 } |
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148 } |
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149 |
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150 int audio_in_start_capture(audio_in_t *ai) |
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151 { |
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152 switch (ai->type) { |
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153 #ifdef HAVE_ALSA9 |
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154 case AUDIO_IN_ALSA: |
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155 return snd_pcm_start(ai->alsa.handle); |
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156 #endif |
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157 case AUDIO_IN_OSS: |
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158 return 0; |
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159 default: |
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160 return -1; |
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161 } |
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162 } |
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163 |
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164 int audio_in_read_chunk(audio_in_t *ai, unsigned char *buffer) |
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165 { |
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166 int ret; |
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167 |
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168 switch (ai->type) { |
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169 #ifdef HAVE_ALSA9 |
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170 case AUDIO_IN_ALSA: |
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171 ret = snd_pcm_readi(ai->alsa.handle, buffer, ai->alsa.chunk_size); |
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172 if (ret != ai->alsa.chunk_size) { |
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173 if (ret < 0) { |
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174 mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", snd_strerror(ret)); |
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175 } else { |
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176 mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); |
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177 } |
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178 return -1; |
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179 } |
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180 return ret; |
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181 #endif |
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182 case AUDIO_IN_OSS: |
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183 ret = read(ai->oss.audio_fd, buffer, ai->blocksize); |
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184 if (ret != ai->blocksize) { |
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185 if (ret < 0) { |
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186 mp_msg(MSGT_TV, MSGL_ERR, "\nerror reading audio: %s\n", strerror(errno)); |
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187 } else { |
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188 mp_msg(MSGT_TV, MSGL_ERR, "\nnot enough audio samples!\n"); |
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189 } |
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190 return -1; |
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191 } |
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192 return ret; |
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193 default: |
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194 return -1; |
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195 } |
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196 } |
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197 |
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198 #endif |