Mercurial > mplayer.hg
annotate libao2/ao_sun.c @ 1393:5eef9e69b145
Beginning sse optimization of dct code.
author | atmos4 |
---|---|
date | Fri, 27 Jul 2001 17:21:23 +0000 |
parents | d6e6132bff35 |
children | 8c57a5a3c645 |
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 audio_fd = -1; |
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58 static enum { |
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59 RTSC_UNKNOWN = 0, |
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60 RTSC_ENABLED, |
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61 RTSC_DISABLED |
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62 } enable_sample_timing; |
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63 |
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64 extern int verbose; |
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65 |
1041 | 66 |
1058 | 67 // convert an OSS audio format specification into a sun audio encoding |
68 static int oss2sunfmt(int oss_format) | |
69 { | |
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70 switch (oss_format){ |
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71 case AFMT_MU_LAW: |
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72 return AUDIO_ENCODING_ULAW; |
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73 case AFMT_A_LAW: |
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74 return AUDIO_ENCODING_ALAW; |
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75 case AFMT_S16_BE: |
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76 case AFMT_S16_LE: |
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77 return AUDIO_ENCODING_LINEAR; |
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78 #ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1... |
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79 case AFMT_U8: |
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80 return AUDIO_ENCODING_LINEAR8; |
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81 #endif |
1058 | 82 #ifdef AUDIO_ENCODING_DVI // Missing on NetBSD... |
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83 case AFMT_IMA_ADPCM: |
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84 return AUDIO_ENCODING_DVI; |
1058 | 85 #endif |
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86 default: |
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87 return AUDIO_ENCODING_NONE; |
1058 | 88 } |
89 } | |
90 | |
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91 // try to figure out, if the soundcard driver provides usable (precise) |
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92 // sample counter information |
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93 static int realtime_samplecounter_available(char *dev) |
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94 { |
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95 int fd = -1; |
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96 audio_info_t info; |
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97 int rtsc_ok = RTSC_DISABLED; |
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98 int len; |
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99 void *silence = NULL; |
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100 struct timeval start, end; |
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101 struct timespec delay; |
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102 int usec_delay; |
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103 unsigned last_samplecnt; |
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104 unsigned increment; |
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105 unsigned min_increment; |
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106 |
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107 len = 44100 * 4 / 4; /* amount of data for 0.25sec of 44.1khz, stereo, |
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108 * 16bit. 44kbyte can be sent to all supported |
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109 * sun audio devices without blocking in the |
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110 * "write" below. |
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111 */ |
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112 silence = calloc(1, len); |
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113 if (silence == NULL) |
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114 goto error; |
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115 |
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116 if ((fd = open(dev, O_WRONLY)) < 0) |
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117 goto error; |
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118 |
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119 AUDIO_INITINFO(&info); |
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120 info.play.sample_rate = 44100; |
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121 info.play.channels = AUDIO_CHANNELS_STEREO; |
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122 info.play.precision = AUDIO_PRECISION_16; |
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123 info.play.encoding = AUDIO_ENCODING_LINEAR; |
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124 info.play.samples = 0; |
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125 if (ioctl(fd, AUDIO_SETINFO, &info)) { |
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126 if (verbose) |
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127 printf("rtsc: SETINFO failed\n"); |
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128 goto error; |
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129 } |
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130 |
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131 if (write(fd, silence, len) != len) { |
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132 if (verbose) |
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133 printf("rtsc: write failed"); |
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134 goto error; |
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135 } |
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136 |
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137 if (ioctl(fd, AUDIO_GETINFO, &info)) { |
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138 if (verbose) |
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139 perror("rtsc: GETINFO1"); |
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140 goto error; |
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141 } |
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142 |
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143 last_samplecnt = info.play.samples; |
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144 min_increment = ~0; |
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145 |
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146 gettimeofday(&start, NULL); |
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147 for (;;) { |
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148 delay.tv_sec = 0; |
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149 delay.tv_nsec = 10000000; |
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150 nanosleep(&delay, NULL); |
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151 gettimeofday(&end, NULL); |
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152 usec_delay = (end.tv_sec - start.tv_sec) * 1000000 |
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153 + end.tv_usec - start.tv_usec; |
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154 |
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155 // stop monitoring sample counter after 0.2 seconds |
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156 if (usec_delay > 200000) |
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157 break; |
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158 |
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159 if (ioctl(fd, AUDIO_GETINFO, &info)) { |
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160 if (verbose) |
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161 perror("rtsc: GETINFO2 failed"); |
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162 goto error; |
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163 } |
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164 if (info.play.samples < last_samplecnt) { |
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165 if (verbose) |
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166 printf("rtsc: %d > %d?\n", last_samplecnt, info.play.samples); |
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167 goto error; |
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168 } |
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169 |
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170 if ((increment = info.play.samples - last_samplecnt) > 0) { |
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171 if (verbose) |
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172 printf("ao_sun: sample counter increment: %d\n", increment); |
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173 if (increment < min_increment) { |
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174 min_increment = increment; |
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175 if (min_increment < 2000) |
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176 break; // looks good |
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177 } |
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178 } |
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179 last_samplecnt = info.play.samples; |
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180 } |
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181 |
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182 /* |
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183 * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes |
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184 * chunks (== 4096 samples) to the audio device. If we see a minimum |
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185 * sample counter increment from the soundcard driver of less than |
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186 * 2000 samples, we assume that the driver provides a useable realtime |
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187 * sample counter in the AUDIO_INFO play.samples field. Timing based |
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188 * on sample counts should be much more accurate than counting whole |
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189 * 16kbyte chunks. |
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190 */ |
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191 if (min_increment < 2000) |
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192 rtsc_ok = RTSC_ENABLED; |
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193 |
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194 if (verbose) |
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195 printf("ao_sun: minimum sample counter increment per 10msec interval: %d\n" |
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196 "\t%susing sample counter based timing code\n", |
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197 min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not "); |
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198 |
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199 |
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200 error: |
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201 if (silence != NULL) free(silence); |
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202 if (fd >= 0) { |
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203 #ifdef __svr4__ |
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204 // remove the 0 bytes from the above measurement from the |
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205 // audio driver's STREAMS queue |
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206 ioctl(fd, I_FLUSH, FLUSHW); |
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207 #endif |
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208 //ioctl(fd, AUDIO_DRAIN, 0); |
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209 close(fd); |
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210 } |
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211 |
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212 return rtsc_ok; |
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213 } |
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214 |
1041 | 215 // to set/get/query special features/parameters |
216 static int control(int cmd,int arg){ | |
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217 switch(cmd){ |
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218 case AOCONTROL_SET_DEVICE: |
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219 audio_dev=(char*)arg; |
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220 return CONTROL_OK; |
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221 case AOCONTROL_QUERY_FORMAT: |
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222 return CONTROL_TRUE; |
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223 } |
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224 return CONTROL_UNKNOWN; |
1041 | 225 } |
226 | |
227 // open & setup audio device | |
228 // return: 1=success 0=fail | |
229 static int init(int rate,int channels,int format,int flags){ | |
230 | |
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231 audio_info_t info; |
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232 int byte_per_sec; |
1041 | 233 |
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234 if (ao_subdevice) audio_dev = ao_subdevice; |
1209 | 235 |
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236 if (enable_sample_timing == RTSC_UNKNOWN |
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237 && !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) { |
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238 enable_sample_timing = realtime_samplecounter_available(audio_dev); |
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239 } |
1041 | 240 |
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241 printf("ao2: %d Hz %d chans %s [0x%X]\n", |
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242 rate,channels,audio_out_format_name(format),format); |
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243 |
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244 audio_fd=open(audio_dev, O_WRONLY); |
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245 if(audio_fd<0){ |
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246 printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno)); |
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247 return 0; |
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248 } |
1041 | 249 |
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250 ioctl(audio_fd, AUDIO_DRAIN, 0); |
1041 | 251 |
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252 AUDIO_INITINFO(&info); |
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253 info.play.encoding = oss2sunfmt(ao_format = format); |
1320
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254 info.play.precision = |
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255 (format==AFMT_S16_LE || format==AFMT_S16_BE |
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256 ? AUDIO_PRECISION_16 |
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257 : AUDIO_PRECISION_8); |
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258 info.play.channels = ao_channels = channels; |
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259 info.play.sample_rate = ao_samplerate = rate; |
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260 if(ioctl (audio_fd, AUDIO_SETINFO, &info)<0) |
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261 printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n", |
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262 channels, audio_out_format_name(format), rate); |
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263 bytes_per_sample = channels * info.play.precision / 8; |
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264 byte_per_sec = bytes_per_sample * rate; |
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265 ao_outburst = byte_per_sec > 100000 ? 16384 : 8192; |
1041 | 266 |
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267 #ifdef __not_used__ |
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268 /* |
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269 * hmm, ao_buffersize is currently not used in this driver, do there's |
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270 * no need to measure it |
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271 */ |
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272 if(ao_buffersize==-1){ |
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273 // Measuring buffer size: |
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274 void* data; |
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275 ao_buffersize=0; |
1041 | 276 #ifdef HAVE_AUDIO_SELECT |
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277 data = malloc(ao_outburst); |
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278 memset(data, format==AFMT_U8 ? 0x80 : 0, ao_outburst); |
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279 while(ao_buffersize<0x40000){ |
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280 fd_set rfds; |
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281 struct timeval tv; |
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282 FD_ZERO(&rfds); FD_SET(audio_fd,&rfds); |
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283 tv.tv_sec=0; tv.tv_usec = 0; |
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284 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break; |
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285 write(audio_fd,data,ao_outburst); |
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286 ao_buffersize+=ao_outburst; |
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287 } |
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288 free(data); |
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289 if(ao_buffersize==0){ |
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290 printf("\n *** Your audio driver DOES NOT support select() ***\n"); |
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291 printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n"); |
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292 return 0; |
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293 } |
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294 #ifdef __svr4__ |
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295 // remove the 0 bytes from the above ao_buffersize measurement from the |
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296 // audio driver's STREAMS queue |
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297 ioctl(audio_fd, I_FLUSH, FLUSHW); |
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298 #endif |
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299 ioctl(audio_fd, AUDIO_DRAIN, 0); |
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300 #endif |
1041 | 301 } |
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302 #endif /* __not_used__ */ |
1041 | 303 |
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304 AUDIO_INITINFO(&info); |
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305 info.play.samples = 0; |
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306 info.play.eof = 0; |
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307 info.play.error = 0; |
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308 ioctl (audio_fd, AUDIO_SETINFO, &info); |
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309 |
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310 queued_bursts = 0; |
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311 queued_samples = 0; |
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312 |
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313 return 1; |
1041 | 314 } |
315 | |
316 // close audio device | |
317 static void uninit(){ | |
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318 #ifdef __svr4__ |
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319 // throw away buffered data in the audio driver's STREAMS queue |
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320 ioctl(audio_fd, I_FLUSH, FLUSHW); |
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321 #endif |
1041 | 322 close(audio_fd); |
323 } | |
324 | |
325 // stop playing and empty buffers (for seeking/pause) | |
326 static void reset(){ | |
327 audio_info_t info; | |
328 | |
329 uninit(); | |
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330 audio_fd=open(audio_dev, O_WRONLY); |
1041 | 331 if(audio_fd<0){ |
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332 printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno)); |
1041 | 333 return; |
334 } | |
335 | |
336 ioctl(audio_fd, AUDIO_DRAIN, 0); | |
337 | |
338 AUDIO_INITINFO(&info); | |
1058 | 339 info.play.encoding = oss2sunfmt(ao_format); |
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340 info.play.precision = |
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341 (ao_format==AFMT_S16_LE || ao_format==AFMT_S16_BE |
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342 ? AUDIO_PRECISION_16 |
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343 : AUDIO_PRECISION_8); |
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344 info.play.channels = ao_channels; |
1041 | 345 info.play.sample_rate = ao_samplerate; |
346 info.play.samples = 0; | |
347 info.play.eof = 0; | |
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348 info.play.error = 0; |
1041 | 349 ioctl (audio_fd, AUDIO_SETINFO, &info); |
350 queued_bursts = 0; | |
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351 queued_samples = 0; |
1041 | 352 } |
353 | |
354 // stop playing, keep buffers (for pause) | |
355 static void audio_pause() | |
356 { | |
357 struct audio_info info; | |
358 AUDIO_INITINFO(&info); | |
359 info.play.pause = 1; | |
360 ioctl(audio_fd, AUDIO_SETINFO, &info); | |
361 } | |
362 | |
363 // resume playing, after audio_pause() | |
364 static void audio_resume() | |
365 { | |
366 struct audio_info info; | |
367 AUDIO_INITINFO(&info); | |
368 info.play.pause = 0; | |
369 ioctl(audio_fd, AUDIO_SETINFO, &info); | |
370 } | |
371 | |
372 | |
373 // return: how many bytes can be played without blocking | |
374 static int get_space(){ | |
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375 int playsize = ao_outburst; |
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376 audio_info_t info; |
1041 | 377 |
378 // check buffer | |
379 #ifdef HAVE_AUDIO_SELECT | |
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380 { |
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381 fd_set rfds; |
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382 struct timeval tv; |
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383 FD_ZERO(&rfds); |
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384 FD_SET(audio_fd, &rfds); |
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385 tv.tv_sec = 0; |
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386 tv.tv_usec = 0; |
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387 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block! |
1041 | 388 } |
389 #endif | |
390 | |
391 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
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392 if (queued_bursts - info.play.eof > 2) |
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393 return 0; |
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394 |
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395 return ao_outburst; |
1041 | 396 } |
397 | |
398 // plays 'len' bytes of 'data' | |
399 // it should round it down to outburst*n | |
400 // return: number of bytes played | |
401 static int play(void* data,int len,int flags){ | |
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402 #if WORDS_BIGENDIAN |
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403 int native_endian = AFMT_S16_BE; |
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404 #else |
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405 int native_endian = AFMT_S16_LE; |
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406 #endif |
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407 |
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408 if (len < ao_outburst) return 0; |
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409 len /= ao_outburst; |
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410 len *= ao_outburst; |
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411 |
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412 /* 16-bit format using the 'wrong' byteorder? swap words */ |
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413 if ((ao_format == AFMT_S16_LE || ao_format == AFMT_S16_BE) |
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414 && ao_format != native_endian) { |
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415 static void *swab_buf; |
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416 static int swab_len; |
1320
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417 if (len > swab_len) { |
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418 if (swab_buf) |
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419 swab_buf = realloc(swab_buf, len); |
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420 else |
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421 swab_buf = malloc(len); |
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422 swab_len = len; |
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423 if (swab_buf == NULL) return 0; |
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424 } |
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425 swab(data, swab_buf, len); |
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426 data = swab_buf; |
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427 } |
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428 |
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429 len = write(audio_fd, data, len); |
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430 if(len > 0) { |
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431 queued_samples += len / bytes_per_sample; |
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432 if (write(audio_fd,data,0) < 0) |
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433 perror("ao_sun: send EOF audio record"); |
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434 else |
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435 queued_bursts ++; |
1041 | 436 } |
437 return len; | |
438 } | |
439 | |
440 | |
441 // return: how many unplayed bytes are in the buffer | |
442 static int get_delay(){ | |
443 audio_info_t info; | |
444 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
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445 if (info.play.samples && enable_sample_timing == RTSC_ENABLED) |
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446 return (queued_samples - info.play.samples) * bytes_per_sample; |
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447 else |
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448 return (queued_bursts - info.play.eof) * ao_outburst; |
1041 | 449 } |
450 |