Mercurial > mplayer.hg
annotate libao2/ao_sun.c @ 1925:633a7f0d72a4
add gui error handling jol.
author | pontscho |
---|---|
date | Thu, 20 Sep 2001 10:27:24 +0000 |
parents | 6da3b9428ff0 |
children | 981a9e5118ce |
rev | line source |
---|---|
1041 | 1 #include <stdio.h> |
2 #include <stdlib.h> | |
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3 #include <string.h> |
1041 | 4 |
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5 #include <unistd.h> |
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6 #include <fcntl.h> |
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7 #include <errno.h> |
1041 | 8 #include <sys/ioctl.h> |
9 #include <sys/time.h> | |
10 #include <sys/types.h> | |
11 #include <sys/stat.h> | |
12 #include <sys/audioio.h> | |
13 #ifdef __svr4__ | |
14 #include <stropts.h> | |
15 #endif | |
16 | |
17 #include "../config.h" | |
18 | |
19 #include "audio_out.h" | |
20 #include "audio_out_internal.h" | |
1058 | 21 #include "afmt.h" |
1041 | 22 |
23 static ao_info_t info = | |
24 { | |
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25 "Sun audio output", |
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26 "sun", |
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27 "jk@tools.de", |
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28 "" |
1041 | 29 }; |
30 | |
31 LIBAO_EXTERN(sun) | |
32 | |
33 | |
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34 /* These defines are missing on NetBSD */ |
1041 | 35 #ifndef AUDIO_PRECISION_8 |
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36 #define AUDIO_PRECISION_8 8 |
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37 #define AUDIO_PRECISION_16 16 |
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38 #endif |
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39 #ifndef AUDIO_CHANNELS_MONO |
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40 #define AUDIO_CHANNELS_MONO 1 |
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41 #define AUDIO_CHANNELS_STEREO 2 |
1041 | 42 #endif |
43 | |
44 | |
45 // there are some globals: | |
46 // ao_samplerate | |
47 // ao_channels | |
48 // ao_format | |
49 // ao_bps | |
50 // ao_outburst | |
51 // ao_buffersize | |
52 | |
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53 static char *audio_dev = "/dev/audio"; |
1041 | 54 static int queued_bursts = 0; |
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55 static int queued_samples = 0; |
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56 static int bytes_per_sample = 0; |
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57 static int convert_u8_s8; |
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58 static int audio_fd = -1; |
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59 static enum { |
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60 RTSC_UNKNOWN = 0, |
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61 RTSC_ENABLED, |
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62 RTSC_DISABLED |
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63 } enable_sample_timing; |
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64 |
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65 extern int verbose; |
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66 |
1041 | 67 |
1058 | 68 // convert an OSS audio format specification into a sun audio encoding |
69 static int oss2sunfmt(int oss_format) | |
70 { | |
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71 switch (oss_format){ |
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72 case AFMT_MU_LAW: |
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73 return AUDIO_ENCODING_ULAW; |
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74 case AFMT_A_LAW: |
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75 return AUDIO_ENCODING_ALAW; |
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76 case AFMT_S16_BE: |
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77 case AFMT_S16_LE: |
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78 return AUDIO_ENCODING_LINEAR; |
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79 #ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1... |
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80 case AFMT_U8: |
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81 return AUDIO_ENCODING_LINEAR8; |
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82 #endif |
1058 | 83 #ifdef AUDIO_ENCODING_DVI // Missing on NetBSD... |
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84 case AFMT_IMA_ADPCM: |
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85 return AUDIO_ENCODING_DVI; |
1058 | 86 #endif |
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87 default: |
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88 return AUDIO_ENCODING_NONE; |
1058 | 89 } |
90 } | |
91 | |
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92 // try to figure out, if the soundcard driver provides usable (precise) |
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93 // sample counter information |
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94 static int realtime_samplecounter_available(char *dev) |
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95 { |
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96 int fd = -1; |
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97 audio_info_t info; |
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98 int rtsc_ok = RTSC_DISABLED; |
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99 int len; |
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100 void *silence = NULL; |
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101 struct timeval start, end; |
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102 struct timespec delay; |
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103 int usec_delay; |
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104 unsigned last_samplecnt; |
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105 unsigned increment; |
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106 unsigned min_increment; |
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107 |
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108 len = 44100 * 4 / 4; /* amount of data for 0.25sec of 44.1khz, stereo, |
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109 * 16bit. 44kbyte can be sent to all supported |
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110 * sun audio devices without blocking in the |
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111 * "write" below. |
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112 */ |
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113 silence = calloc(1, len); |
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114 if (silence == NULL) |
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115 goto error; |
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116 |
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117 if ((fd = open(dev, O_WRONLY)) < 0) |
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118 goto error; |
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119 |
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120 AUDIO_INITINFO(&info); |
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121 info.play.sample_rate = 44100; |
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122 info.play.channels = AUDIO_CHANNELS_STEREO; |
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123 info.play.precision = AUDIO_PRECISION_16; |
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124 info.play.encoding = AUDIO_ENCODING_LINEAR; |
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125 info.play.samples = 0; |
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126 if (ioctl(fd, AUDIO_SETINFO, &info)) { |
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127 if (verbose) |
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128 printf("rtsc: SETINFO failed\n"); |
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129 goto error; |
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130 } |
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131 |
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132 if (write(fd, silence, len) != len) { |
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133 if (verbose) |
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134 printf("rtsc: write failed"); |
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135 goto error; |
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136 } |
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137 |
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138 if (ioctl(fd, AUDIO_GETINFO, &info)) { |
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139 if (verbose) |
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140 perror("rtsc: GETINFO1"); |
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141 goto error; |
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142 } |
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143 |
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144 last_samplecnt = info.play.samples; |
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145 min_increment = ~0; |
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146 |
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147 gettimeofday(&start, NULL); |
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148 for (;;) { |
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149 delay.tv_sec = 0; |
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150 delay.tv_nsec = 10000000; |
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151 nanosleep(&delay, NULL); |
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152 gettimeofday(&end, NULL); |
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153 usec_delay = (end.tv_sec - start.tv_sec) * 1000000 |
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154 + end.tv_usec - start.tv_usec; |
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155 |
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156 // stop monitoring sample counter after 0.2 seconds |
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157 if (usec_delay > 200000) |
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158 break; |
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159 |
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160 if (ioctl(fd, AUDIO_GETINFO, &info)) { |
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161 if (verbose) |
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162 perror("rtsc: GETINFO2 failed"); |
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163 goto error; |
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164 } |
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165 if (info.play.samples < last_samplecnt) { |
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166 if (verbose) |
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167 printf("rtsc: %d > %d?\n", last_samplecnt, info.play.samples); |
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168 goto error; |
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169 } |
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170 |
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171 if ((increment = info.play.samples - last_samplecnt) > 0) { |
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172 if (verbose) |
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173 printf("ao_sun: sample counter increment: %d\n", increment); |
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174 if (increment < min_increment) { |
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175 min_increment = increment; |
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176 if (min_increment < 2000) |
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177 break; // looks good |
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178 } |
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179 } |
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180 last_samplecnt = info.play.samples; |
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181 } |
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182 |
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183 /* |
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184 * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes |
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185 * chunks (== 4096 samples) to the audio device. If we see a minimum |
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186 * sample counter increment from the soundcard driver of less than |
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187 * 2000 samples, we assume that the driver provides a useable realtime |
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188 * sample counter in the AUDIO_INFO play.samples field. Timing based |
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189 * on sample counts should be much more accurate than counting whole |
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190 * 16kbyte chunks. |
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191 */ |
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192 if (min_increment < 2000) |
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193 rtsc_ok = RTSC_ENABLED; |
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194 |
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195 if (verbose) |
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196 printf("ao_sun: minimum sample counter increment per 10msec interval: %d\n" |
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197 "\t%susing sample counter based timing code\n", |
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198 min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not "); |
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199 |
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200 |
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201 error: |
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202 if (silence != NULL) free(silence); |
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203 if (fd >= 0) { |
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204 #ifdef __svr4__ |
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205 // remove the 0 bytes from the above measurement from the |
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206 // audio driver's STREAMS queue |
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207 ioctl(fd, I_FLUSH, FLUSHW); |
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208 #endif |
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209 //ioctl(fd, AUDIO_DRAIN, 0); |
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210 close(fd); |
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211 } |
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212 |
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213 return rtsc_ok; |
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214 } |
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215 |
1041 | 216 // to set/get/query special features/parameters |
217 static int control(int cmd,int arg){ | |
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218 switch(cmd){ |
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219 case AOCONTROL_SET_DEVICE: |
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220 audio_dev=(char*)arg; |
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221 return CONTROL_OK; |
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222 case AOCONTROL_QUERY_FORMAT: |
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223 return CONTROL_TRUE; |
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224 } |
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225 return CONTROL_UNKNOWN; |
1041 | 226 } |
227 | |
228 // open & setup audio device | |
229 // return: 1=success 0=fail | |
230 static int init(int rate,int channels,int format,int flags){ | |
231 | |
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232 audio_info_t info; |
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233 int byte_per_sec; |
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234 int ok; |
1041 | 235 |
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236 if (ao_subdevice) audio_dev = ao_subdevice; |
1209 | 237 |
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238 if (enable_sample_timing == RTSC_UNKNOWN |
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239 && !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) { |
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240 enable_sample_timing = realtime_samplecounter_available(audio_dev); |
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241 } |
1041 | 242 |
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243 // printf("ao2: %d Hz %d chans %s [0x%X]\n", |
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244 // rate,channels,audio_out_format_name(format),format); |
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245 |
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246 audio_fd=open(audio_dev, O_WRONLY); |
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247 if(audio_fd<0){ |
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248 printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno)); |
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249 return 0; |
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250 } |
1041 | 251 |
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252 ioctl(audio_fd, AUDIO_DRAIN, 0); |
1041 | 253 |
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254 AUDIO_INITINFO(&info); |
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255 info.play.encoding = oss2sunfmt(ao_format = format); |
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256 info.play.precision = |
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257 (format==AFMT_S16_LE || format==AFMT_S16_BE |
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258 ? AUDIO_PRECISION_16 |
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259 : AUDIO_PRECISION_8); |
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260 info.play.channels = ao_channels = channels; |
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261 info.play.sample_rate = ao_samplerate = rate; |
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262 convert_u8_s8 = 0; |
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263 ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0; |
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264 if (!ok && info.play.encoding == AUDIO_ENCODING_LINEAR8) { |
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265 /* sun audiocs hardware does not support U8 format, try S8... */ |
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266 info.play.encoding = AUDIO_ENCODING_LINEAR; |
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267 ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0; |
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268 if (ok) { |
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269 /* we must perform software U8 -> S8 conversion */ |
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270 convert_u8_s8 = 1; |
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271 } |
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272 } |
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273 if (!ok) { |
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274 printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n", |
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275 channels, audio_out_format_name(format), rate); |
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276 return 0; |
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277 } |
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278 |
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279 bytes_per_sample = channels * info.play.precision / 8; |
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280 byte_per_sec = bytes_per_sample * rate; |
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|
281 ao_outburst = byte_per_sec > 100000 ? 16384 : 8192; |
1041 | 282 |
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283 #ifdef __not_used__ |
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284 /* |
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285 * hmm, ao_buffersize is currently not used in this driver, do there's |
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286 * no need to measure it |
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287 */ |
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288 if(ao_buffersize==-1){ |
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289 // Measuring buffer size: |
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290 void* data; |
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291 ao_buffersize=0; |
1041 | 292 #ifdef HAVE_AUDIO_SELECT |
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293 data = malloc(ao_outburst); |
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294 memset(data, format==AFMT_U8 ? 0x80 : 0, ao_outburst); |
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295 while(ao_buffersize<0x40000){ |
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296 fd_set rfds; |
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297 struct timeval tv; |
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298 FD_ZERO(&rfds); FD_SET(audio_fd,&rfds); |
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299 tv.tv_sec=0; tv.tv_usec = 0; |
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300 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break; |
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301 write(audio_fd,data,ao_outburst); |
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302 ao_buffersize+=ao_outburst; |
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303 } |
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304 free(data); |
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305 if(ao_buffersize==0){ |
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306 printf("\n *** Your audio driver DOES NOT support select() ***\n"); |
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307 printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n"); |
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308 return 0; |
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309 } |
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310 #ifdef __svr4__ |
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311 // remove the 0 bytes from the above ao_buffersize measurement from the |
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312 // audio driver's STREAMS queue |
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313 ioctl(audio_fd, I_FLUSH, FLUSHW); |
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314 #endif |
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315 ioctl(audio_fd, AUDIO_DRAIN, 0); |
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316 #endif |
1041 | 317 } |
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318 #endif /* __not_used__ */ |
1041 | 319 |
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320 AUDIO_INITINFO(&info); |
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321 info.play.samples = 0; |
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322 info.play.eof = 0; |
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323 info.play.error = 0; |
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324 ioctl (audio_fd, AUDIO_SETINFO, &info); |
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325 |
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326 queued_bursts = 0; |
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327 queued_samples = 0; |
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328 |
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329 return 1; |
1041 | 330 } |
331 | |
332 // close audio device | |
333 static void uninit(){ | |
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334 #ifdef __svr4__ |
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335 // throw away buffered data in the audio driver's STREAMS queue |
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336 ioctl(audio_fd, I_FLUSH, FLUSHW); |
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337 #endif |
1041 | 338 close(audio_fd); |
339 } | |
340 | |
341 // stop playing and empty buffers (for seeking/pause) | |
342 static void reset(){ | |
343 audio_info_t info; | |
344 | |
345 uninit(); | |
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346 audio_fd=open(audio_dev, O_WRONLY); |
1041 | 347 if(audio_fd<0){ |
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348 printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno)); |
1041 | 349 return; |
350 } | |
351 | |
352 ioctl(audio_fd, AUDIO_DRAIN, 0); | |
353 | |
354 AUDIO_INITINFO(&info); | |
1058 | 355 info.play.encoding = oss2sunfmt(ao_format); |
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356 info.play.precision = |
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357 (ao_format==AFMT_S16_LE || ao_format==AFMT_S16_BE |
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358 ? AUDIO_PRECISION_16 |
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359 : AUDIO_PRECISION_8); |
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360 info.play.channels = ao_channels; |
1041 | 361 info.play.sample_rate = ao_samplerate; |
362 info.play.samples = 0; | |
363 info.play.eof = 0; | |
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364 info.play.error = 0; |
1041 | 365 ioctl (audio_fd, AUDIO_SETINFO, &info); |
366 queued_bursts = 0; | |
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367 queued_samples = 0; |
1041 | 368 } |
369 | |
370 // stop playing, keep buffers (for pause) | |
371 static void audio_pause() | |
372 { | |
373 struct audio_info info; | |
374 AUDIO_INITINFO(&info); | |
375 info.play.pause = 1; | |
376 ioctl(audio_fd, AUDIO_SETINFO, &info); | |
377 } | |
378 | |
379 // resume playing, after audio_pause() | |
380 static void audio_resume() | |
381 { | |
382 struct audio_info info; | |
383 AUDIO_INITINFO(&info); | |
384 info.play.pause = 0; | |
385 ioctl(audio_fd, AUDIO_SETINFO, &info); | |
386 } | |
387 | |
388 | |
389 // return: how many bytes can be played without blocking | |
390 static int get_space(){ | |
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391 int playsize = ao_outburst; |
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392 audio_info_t info; |
1041 | 393 |
394 // check buffer | |
395 #ifdef HAVE_AUDIO_SELECT | |
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396 { |
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397 fd_set rfds; |
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398 struct timeval tv; |
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399 FD_ZERO(&rfds); |
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400 FD_SET(audio_fd, &rfds); |
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401 tv.tv_sec = 0; |
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402 tv.tv_usec = 0; |
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403 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block! |
1041 | 404 } |
405 #endif | |
406 | |
407 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
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408 if (queued_bursts - info.play.eof > 2) |
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409 return 0; |
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410 |
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411 return ao_outburst; |
1041 | 412 } |
413 | |
414 // plays 'len' bytes of 'data' | |
415 // it should round it down to outburst*n | |
416 // return: number of bytes played | |
417 static int play(void* data,int len,int flags){ | |
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418 #if WORDS_BIGENDIAN |
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419 int native_endian = AFMT_S16_BE; |
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420 #else |
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421 int native_endian = AFMT_S16_LE; |
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422 #endif |
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423 |
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424 if (len < ao_outburst) return 0; |
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425 len /= ao_outburst; |
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426 len *= ao_outburst; |
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427 |
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428 /* 16-bit format using the 'wrong' byteorder? swap words */ |
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429 if ((ao_format == AFMT_S16_LE || ao_format == AFMT_S16_BE) |
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430 && ao_format != native_endian) { |
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431 static void *swab_buf; |
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432 static int swab_len; |
1320
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433 if (len > swab_len) { |
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434 if (swab_buf) |
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435 swab_buf = realloc(swab_buf, len); |
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436 else |
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437 swab_buf = malloc(len); |
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438 swab_len = len; |
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439 if (swab_buf == NULL) return 0; |
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440 } |
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441 swab(data, swab_buf, len); |
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442 data = swab_buf; |
1807
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443 } else if (ao_format == AFMT_U8 && convert_u8_s8) { |
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444 int i; |
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445 unsigned char *p = data; |
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446 |
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447 for (i = 0, p = data; i < len; i++, p++) |
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448 *p ^= 0x80; |
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449 } |
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450 |
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451 len = write(audio_fd, data, len); |
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452 if(len > 0) { |
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453 queued_samples += len / bytes_per_sample; |
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454 if (write(audio_fd,data,0) < 0) |
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455 perror("ao_sun: send EOF audio record"); |
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jkeil
parents:
1209
diff
changeset
|
456 else |
ffd63a75700c
Support playback of AFMT_S16_LE audio data on a big endian machine
jkeil
parents:
1209
diff
changeset
|
457 queued_bursts ++; |
1041 | 458 } |
459 return len; | |
460 } | |
461 | |
462 | |
463 // return: how many unplayed bytes are in the buffer | |
464 static int get_delay(){ | |
465 audio_info_t info; | |
466 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
1196
fc51929ec8ea
Applied patch by J¸«ärgen Keil (jk@tools.de), improves smoothness of video playback.
atmosfear
parents:
1058
diff
changeset
|
467 if (info.play.samples && enable_sample_timing == RTSC_ENABLED) |
fc51929ec8ea
Applied patch by J¸«ärgen Keil (jk@tools.de), improves smoothness of video playback.
atmosfear
parents:
1058
diff
changeset
|
468 return (queued_samples - info.play.samples) * bytes_per_sample; |
fc51929ec8ea
Applied patch by J¸«ärgen Keil (jk@tools.de), improves smoothness of video playback.
atmosfear
parents:
1058
diff
changeset
|
469 else |
fc51929ec8ea
Applied patch by J¸«ärgen Keil (jk@tools.de), improves smoothness of video playback.
atmosfear
parents:
1058
diff
changeset
|
470 return (queued_bursts - info.play.eof) * ao_outburst; |
1041 | 471 } |
472 |