Mercurial > mplayer.hg
annotate libao2/ao_sun.c @ 4302:9f12fd5f47d0
Modified the cache option value setting/reading, so we can report
in any case the cache size value used.
author | bertrand |
---|---|
date | Tue, 22 Jan 2002 06:12:18 +0000 |
parents | 24b0fad7fccc |
children | d678ce495a75 |
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 | |
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45 static char *audio_dev = NULL; |
1041 | 46 static int queued_bursts = 0; |
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47 static int queued_samples = 0; |
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48 static int bytes_per_sample = 0; |
3095 | 49 static int byte_per_sec = 0; |
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50 static int convert_u8_s8; |
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51 static int audio_fd = -1; |
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52 static enum { |
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53 RTSC_UNKNOWN = 0, |
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54 RTSC_ENABLED, |
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55 RTSC_DISABLED |
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56 } enable_sample_timing; |
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57 |
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58 extern int verbose; |
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59 |
1041 | 60 |
1058 | 61 // convert an OSS audio format specification into a sun audio encoding |
62 static int oss2sunfmt(int oss_format) | |
63 { | |
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64 switch (oss_format){ |
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65 case AFMT_MU_LAW: |
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66 return AUDIO_ENCODING_ULAW; |
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67 case AFMT_A_LAW: |
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68 return AUDIO_ENCODING_ALAW; |
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69 case AFMT_S16_BE: |
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70 case AFMT_S16_LE: |
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71 return AUDIO_ENCODING_LINEAR; |
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72 #ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1... |
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73 case AFMT_U8: |
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74 return AUDIO_ENCODING_LINEAR8; |
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75 #endif |
1058 | 76 #ifdef AUDIO_ENCODING_DVI // Missing on NetBSD... |
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77 case AFMT_IMA_ADPCM: |
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78 return AUDIO_ENCODING_DVI; |
1058 | 79 #endif |
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80 default: |
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81 return AUDIO_ENCODING_NONE; |
1058 | 82 } |
83 } | |
84 | |
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85 // try to figure out, if the soundcard driver provides usable (precise) |
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86 // sample counter information |
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87 static int realtime_samplecounter_available(char *dev) |
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88 { |
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89 int fd = -1; |
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90 audio_info_t info; |
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91 int rtsc_ok = RTSC_DISABLED; |
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92 int len; |
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93 void *silence = NULL; |
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94 struct timeval start, end; |
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95 struct timespec delay; |
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96 int usec_delay; |
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97 unsigned last_samplecnt; |
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98 unsigned increment; |
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99 unsigned min_increment; |
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100 |
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101 len = 44100 * 4 / 4; /* amount of data for 0.25sec of 44.1khz, stereo, |
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102 * 16bit. 44kbyte can be sent to all supported |
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103 * sun audio devices without blocking in the |
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104 * "write" below. |
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105 */ |
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106 silence = calloc(1, len); |
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107 if (silence == NULL) |
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108 goto error; |
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109 |
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110 if ((fd = open(dev, O_WRONLY)) < 0) |
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111 goto error; |
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112 |
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113 AUDIO_INITINFO(&info); |
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114 info.play.sample_rate = 44100; |
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115 info.play.channels = AUDIO_CHANNELS_STEREO; |
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116 info.play.precision = AUDIO_PRECISION_16; |
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117 info.play.encoding = AUDIO_ENCODING_LINEAR; |
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118 info.play.samples = 0; |
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119 if (ioctl(fd, AUDIO_SETINFO, &info)) { |
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120 if (verbose) |
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121 printf("rtsc: SETINFO failed\n"); |
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122 goto error; |
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123 } |
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124 |
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125 if (write(fd, silence, len) != len) { |
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126 if (verbose) |
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127 printf("rtsc: write failed"); |
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128 goto error; |
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129 } |
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130 |
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131 if (ioctl(fd, AUDIO_GETINFO, &info)) { |
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132 if (verbose) |
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133 perror("rtsc: GETINFO1"); |
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134 goto error; |
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135 } |
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136 |
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137 last_samplecnt = info.play.samples; |
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138 min_increment = ~0; |
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139 |
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140 gettimeofday(&start, NULL); |
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141 for (;;) { |
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142 delay.tv_sec = 0; |
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143 delay.tv_nsec = 10000000; |
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144 nanosleep(&delay, NULL); |
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145 gettimeofday(&end, NULL); |
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146 usec_delay = (end.tv_sec - start.tv_sec) * 1000000 |
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147 + end.tv_usec - start.tv_usec; |
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148 |
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149 // stop monitoring sample counter after 0.2 seconds |
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150 if (usec_delay > 200000) |
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151 break; |
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152 |
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153 if (ioctl(fd, AUDIO_GETINFO, &info)) { |
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154 if (verbose) |
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155 perror("rtsc: GETINFO2 failed"); |
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156 goto error; |
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157 } |
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158 if (info.play.samples < last_samplecnt) { |
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159 if (verbose) |
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160 printf("rtsc: %d > %d?\n", last_samplecnt, info.play.samples); |
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161 goto error; |
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162 } |
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163 |
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164 if ((increment = info.play.samples - last_samplecnt) > 0) { |
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165 if (verbose) |
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166 printf("ao_sun: sample counter increment: %d\n", increment); |
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167 if (increment < min_increment) { |
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168 min_increment = increment; |
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169 if (min_increment < 2000) |
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170 break; // looks good |
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171 } |
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172 } |
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173 last_samplecnt = info.play.samples; |
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174 } |
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175 |
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176 /* |
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177 * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes |
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178 * chunks (== 4096 samples) to the audio device. If we see a minimum |
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179 * sample counter increment from the soundcard driver of less than |
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180 * 2000 samples, we assume that the driver provides a useable realtime |
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181 * sample counter in the AUDIO_INFO play.samples field. Timing based |
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182 * on sample counts should be much more accurate than counting whole |
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183 * 16kbyte chunks. |
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184 */ |
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185 if (min_increment < 2000) |
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186 rtsc_ok = RTSC_ENABLED; |
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187 |
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188 if (verbose) |
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189 printf("ao_sun: minimum sample counter increment per 10msec interval: %d\n" |
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190 "\t%susing sample counter based timing code\n", |
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191 min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not "); |
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192 |
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193 |
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194 error: |
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195 if (silence != NULL) free(silence); |
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196 if (fd >= 0) { |
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197 #ifdef __svr4__ |
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198 // remove the 0 bytes from the above measurement from the |
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199 // audio driver's STREAMS queue |
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200 ioctl(fd, I_FLUSH, FLUSHW); |
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201 #endif |
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202 //ioctl(fd, AUDIO_DRAIN, 0); |
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203 close(fd); |
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204 } |
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205 |
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206 return rtsc_ok; |
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207 } |
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208 |
1041 | 209 // to set/get/query special features/parameters |
210 static int control(int cmd,int arg){ | |
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211 switch(cmd){ |
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212 case AOCONTROL_SET_DEVICE: |
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213 audio_dev=(char*)arg; |
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214 return CONTROL_OK; |
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215 case AOCONTROL_QUERY_FORMAT: |
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216 return CONTROL_TRUE; |
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217 } |
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218 return CONTROL_UNKNOWN; |
1041 | 219 } |
220 | |
221 // open & setup audio device | |
222 // return: 1=success 0=fail | |
223 static int init(int rate,int channels,int format,int flags){ | |
224 | |
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225 audio_info_t info; |
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226 int ok; |
1041 | 227 |
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228 if (audio_dev == NULL) { |
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229 if ((audio_dev = getenv("AUDIODEV")) == NULL) |
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230 audio_dev = "/dev/audio"; |
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231 } |
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232 |
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233 if (ao_subdevice) audio_dev = ao_subdevice; |
1209 | 234 |
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235 if (enable_sample_timing == RTSC_UNKNOWN |
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236 && !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) { |
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237 enable_sample_timing = realtime_samplecounter_available(audio_dev); |
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238 } |
1041 | 239 |
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240 // printf("ao2: %d Hz %d chans %s [0x%X]\n", |
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241 // rate,channels,audio_out_format_name(format),format); |
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242 |
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243 audio_fd=open(audio_dev, O_WRONLY); |
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244 if(audio_fd<0){ |
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245 printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno)); |
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246 return 0; |
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247 } |
1041 | 248 |
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249 ioctl(audio_fd, AUDIO_DRAIN, 0); |
1041 | 250 |
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251 AUDIO_INITINFO(&info); |
3095 | 252 info.play.encoding = oss2sunfmt(ao_data.format = format); |
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253 info.play.precision = |
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254 (format==AFMT_S16_LE || format==AFMT_S16_BE |
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255 ? AUDIO_PRECISION_16 |
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256 : AUDIO_PRECISION_8); |
3095 | 257 info.play.channels = ao_data.channels = channels; |
258 info.play.sample_rate = ao_data.samplerate = rate; | |
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259 convert_u8_s8 = 0; |
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260 ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0; |
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261 if (!ok && info.play.encoding == AUDIO_ENCODING_LINEAR8) { |
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262 /* sun audiocs hardware does not support U8 format, try S8... */ |
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263 info.play.encoding = AUDIO_ENCODING_LINEAR; |
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264 ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0; |
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265 if (ok) { |
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266 /* we must perform software U8 -> S8 conversion */ |
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267 convert_u8_s8 = 1; |
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268 } |
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269 } |
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|
270 if (!ok) { |
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271 printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n", |
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|
272 channels, audio_out_format_name(format), rate); |
1807
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273 return 0; |
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274 } |
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275 |
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276 bytes_per_sample = channels * info.play.precision / 8; |
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277 byte_per_sec = bytes_per_sample * rate; |
3095 | 278 ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192; |
1041 | 279 |
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|
280 #ifdef __not_used__ |
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281 /* |
3095 | 282 * hmm, ao_data.buffersize is currently not used in this driver, do there's |
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283 * no need to measure it |
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284 */ |
3095 | 285 if(ao_data.buffersize==-1){ |
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286 // Measuring buffer size: |
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287 void* data; |
3095 | 288 ao_data.buffersize=0; |
1041 | 289 #ifdef HAVE_AUDIO_SELECT |
3095 | 290 data = malloc(ao_data.outburst); |
291 memset(data, format==AFMT_U8 ? 0x80 : 0, ao_data.outburst); | |
292 while(ao_data.buffersize<0x40000){ | |
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293 fd_set rfds; |
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294 struct timeval tv; |
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295 FD_ZERO(&rfds); FD_SET(audio_fd,&rfds); |
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296 tv.tv_sec=0; tv.tv_usec = 0; |
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297 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break; |
3095 | 298 write(audio_fd,data,ao_data.outburst); |
299 ao_data.buffersize+=ao_data.outburst; | |
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300 } |
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301 free(data); |
3095 | 302 if(ao_data.buffersize==0){ |
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303 printf("\n *** Your audio driver DOES NOT support select() ***\n"); |
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304 printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n"); |
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305 return 0; |
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306 } |
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307 #ifdef __svr4__ |
3095 | 308 // remove the 0 bytes from the above ao_data.buffersize measurement from the |
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309 // audio driver's STREAMS queue |
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310 ioctl(audio_fd, I_FLUSH, FLUSHW); |
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311 #endif |
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312 ioctl(audio_fd, AUDIO_DRAIN, 0); |
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313 #endif |
1041 | 314 } |
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315 #endif /* __not_used__ */ |
1041 | 316 |
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317 AUDIO_INITINFO(&info); |
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318 info.play.samples = 0; |
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319 info.play.eof = 0; |
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320 info.play.error = 0; |
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321 ioctl (audio_fd, AUDIO_SETINFO, &info); |
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322 |
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323 queued_bursts = 0; |
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324 queued_samples = 0; |
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325 |
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326 return 1; |
1041 | 327 } |
328 | |
329 // close audio device | |
330 static void uninit(){ | |
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331 #ifdef __svr4__ |
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332 // throw away buffered data in the audio driver's STREAMS queue |
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333 ioctl(audio_fd, I_FLUSH, FLUSHW); |
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334 #endif |
1041 | 335 close(audio_fd); |
336 } | |
337 | |
338 // stop playing and empty buffers (for seeking/pause) | |
339 static void reset(){ | |
340 audio_info_t info; | |
341 | |
342 uninit(); | |
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343 audio_fd=open(audio_dev, O_WRONLY); |
1041 | 344 if(audio_fd<0){ |
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345 printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno)); |
1041 | 346 return; |
347 } | |
348 | |
349 ioctl(audio_fd, AUDIO_DRAIN, 0); | |
350 | |
351 AUDIO_INITINFO(&info); | |
3095 | 352 info.play.encoding = oss2sunfmt(ao_data.format); |
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353 info.play.precision = |
3095 | 354 (ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_S16_BE |
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355 ? AUDIO_PRECISION_16 |
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356 : AUDIO_PRECISION_8); |
3095 | 357 info.play.channels = ao_data.channels; |
358 info.play.sample_rate = ao_data.samplerate; | |
1041 | 359 info.play.samples = 0; |
360 info.play.eof = 0; | |
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361 info.play.error = 0; |
1041 | 362 ioctl (audio_fd, AUDIO_SETINFO, &info); |
363 queued_bursts = 0; | |
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364 queued_samples = 0; |
1041 | 365 } |
366 | |
367 // stop playing, keep buffers (for pause) | |
368 static void audio_pause() | |
369 { | |
370 struct audio_info info; | |
371 AUDIO_INITINFO(&info); | |
372 info.play.pause = 1; | |
373 ioctl(audio_fd, AUDIO_SETINFO, &info); | |
374 } | |
375 | |
376 // resume playing, after audio_pause() | |
377 static void audio_resume() | |
378 { | |
379 struct audio_info info; | |
380 AUDIO_INITINFO(&info); | |
381 info.play.pause = 0; | |
382 ioctl(audio_fd, AUDIO_SETINFO, &info); | |
383 } | |
384 | |
385 | |
386 // return: how many bytes can be played without blocking | |
387 static int get_space(){ | |
3095 | 388 int playsize = ao_data.outburst; |
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389 audio_info_t info; |
1041 | 390 |
391 // check buffer | |
392 #ifdef HAVE_AUDIO_SELECT | |
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393 { |
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394 fd_set rfds; |
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395 struct timeval tv; |
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396 FD_ZERO(&rfds); |
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397 FD_SET(audio_fd, &rfds); |
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398 tv.tv_sec = 0; |
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399 tv.tv_usec = 0; |
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400 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block! |
1041 | 401 } |
402 #endif | |
403 | |
404 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
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405 if (queued_bursts - info.play.eof > 2) |
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406 return 0; |
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407 |
3095 | 408 return ao_data.outburst; |
1041 | 409 } |
410 | |
411 // plays 'len' bytes of 'data' | |
412 // it should round it down to outburst*n | |
413 // return: number of bytes played | |
414 static int play(void* data,int len,int flags){ | |
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415 #if WORDS_BIGENDIAN |
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416 int native_endian = AFMT_S16_BE; |
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417 #else |
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418 int native_endian = AFMT_S16_LE; |
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419 #endif |
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420 |
3095 | 421 if (len < ao_data.outburst) return 0; |
422 len /= ao_data.outburst; | |
423 len *= ao_data.outburst; | |
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424 |
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425 /* 16-bit format using the 'wrong' byteorder? swap words */ |
3095 | 426 if ((ao_data.format == AFMT_S16_LE || ao_data.format == AFMT_S16_BE) |
427 && ao_data.format != native_endian) { | |
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428 static void *swab_buf; |
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429 static int swab_len; |
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430 if (len > swab_len) { |
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431 if (swab_buf) |
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432 swab_buf = realloc(swab_buf, len); |
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433 else |
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434 swab_buf = malloc(len); |
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435 swab_len = len; |
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436 if (swab_buf == NULL) return 0; |
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437 } |
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438 swab(data, swab_buf, len); |
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439 data = swab_buf; |
3095 | 440 } else if (ao_data.format == AFMT_U8 && convert_u8_s8) { |
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441 int i; |
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442 unsigned char *p = data; |
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443 |
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444 for (i = 0, p = data; i < len; i++, p++) |
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445 *p ^= 0x80; |
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446 } |
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447 |
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448 len = write(audio_fd, data, len); |
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449 if(len > 0) { |
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450 queued_samples += len / bytes_per_sample; |
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451 if (write(audio_fd,data,0) < 0) |
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452 perror("ao_sun: send EOF audio record"); |
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453 else |
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454 queued_bursts ++; |
1041 | 455 } |
456 return len; | |
457 } | |
458 | |
459 | |
3095 | 460 // return: delay in seconds between first and last sample in buffer |
461 static float get_delay(){ | |
1041 | 462 audio_info_t info; |
463 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
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464 if (info.play.samples && enable_sample_timing == RTSC_ENABLED) |
3095 | 465 return (float)(queued_samples - info.play.samples) / (float)byte_per_sec; |
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466 else |
3137 | 467 return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec; |
1041 | 468 } |
469 |