annotate libmpdemux/audio_in.c @ 11825:38887cab0d8b

How come TODO lists never shrink?
author diego
date Wed, 21 Jan 2004 21:20:41 +0000
parents 66e491c35dc8
children dfbe8cd0e081
Ignore whitespace changes - Everywhere: Within whitespace: At end of lines:
rev   line source
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13fcab6fde41 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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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) || 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) {
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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) {
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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) {
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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) {
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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 {
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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) {
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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) {
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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) {
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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) {
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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
13fcab6fde41 audio in layer is used only bu tvi_v4l, don't compile for others.
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222 #endif