Mercurial > mplayer.hg
annotate stream/audio_in.c @ 19285:87e3e034e324
Ignore .depend and stream.a.
author | diego |
---|---|
date | Tue, 01 Aug 2006 15:54:23 +0000 |
parents | 64d82a45a05d |
children | e7c989f7a7c9 |
rev | line source |
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7213
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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 #include "audio_in.h" |
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8 #include "mp_msg.h" |
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libmpdemux translatables to help_mp part 1 / mp_msg calls / try 2
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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) { |
11775 | 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) { |
11775 | 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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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) { |
11775 | 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 { |
11775 | 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) { |
11775 | 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) { |
11775 | 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) { |
11775 | 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) { |
11775 | 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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208 mp_msg(MSGT_TV, MSGL_ERR, MSGTR_MPDEMUX_AUDIOIN_ErrReadingAudio, strerror(errno)); |
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209 } else { |
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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; |
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215 #endif |
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216 default: |
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217 return -1; |
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218 } |
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219 } |