Mercurial > mplayer.hg
annotate libao2/ao_sun.c @ 6540:5b3cace64e9d
100l (non mmx versions didnt compile)
some cleanup
author | michael |
---|---|
date | Mon, 24 Jun 2002 00:18:29 +0000 |
parents | 6a6107da2d82 |
children | daf0d43ccde2 |
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" | |
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Moved HW dependent mixer stuff to libao and removed master switch
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18 #include "../mixer.h" |
1041 | 19 |
20 #include "audio_out.h" | |
21 #include "audio_out_internal.h" | |
1058 | 22 #include "afmt.h" |
1041 | 23 |
24 static ao_info_t info = | |
25 { | |
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26 "Sun audio output", |
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27 "sun", |
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28 "jk@tools.de", |
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29 "" |
1041 | 30 }; |
31 | |
32 LIBAO_EXTERN(sun) | |
33 | |
34 | |
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35 /* These defines are missing on NetBSD */ |
1041 | 36 #ifndef AUDIO_PRECISION_8 |
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37 #define AUDIO_PRECISION_8 8 |
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38 #define AUDIO_PRECISION_16 16 |
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39 #endif |
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40 #ifndef AUDIO_CHANNELS_MONO |
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41 #define AUDIO_CHANNELS_MONO 1 |
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42 #define AUDIO_CHANNELS_STEREO 2 |
1041 | 43 #endif |
44 | |
45 | |
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46 static char *sun_mixer_device = NULL; |
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47 static char *audio_dev = NULL; |
1041 | 48 static int queued_bursts = 0; |
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49 static int queued_samples = 0; |
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50 static int bytes_per_sample = 0; |
3095 | 51 static int byte_per_sec = 0; |
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52 static int convert_u8_s8; |
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53 static int audio_fd = -1; |
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54 static enum { |
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55 RTSC_UNKNOWN = 0, |
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56 RTSC_ENABLED, |
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57 RTSC_DISABLED |
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58 } enable_sample_timing; |
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59 |
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60 extern int verbose; |
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61 |
1041 | 62 |
1058 | 63 // convert an OSS audio format specification into a sun audio encoding |
64 static int oss2sunfmt(int oss_format) | |
65 { | |
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66 switch (oss_format){ |
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67 case AFMT_MU_LAW: |
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68 return AUDIO_ENCODING_ULAW; |
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69 case AFMT_A_LAW: |
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70 return AUDIO_ENCODING_ALAW; |
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71 case AFMT_S16_BE: |
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72 case AFMT_S16_LE: |
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73 return AUDIO_ENCODING_LINEAR; |
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74 #ifdef AUDIO_ENCODING_LINEAR8 // Missing on SunOS 5.5.1... |
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75 case AFMT_U8: |
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76 return AUDIO_ENCODING_LINEAR8; |
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77 #endif |
1058 | 78 #ifdef AUDIO_ENCODING_DVI // Missing on NetBSD... |
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79 case AFMT_IMA_ADPCM: |
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80 return AUDIO_ENCODING_DVI; |
1058 | 81 #endif |
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82 default: |
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83 return AUDIO_ENCODING_NONE; |
1058 | 84 } |
85 } | |
86 | |
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87 // try to figure out, if the soundcard driver provides usable (precise) |
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88 // sample counter information |
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89 static int realtime_samplecounter_available(char *dev) |
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90 { |
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91 int fd = -1; |
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92 audio_info_t info; |
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93 int rtsc_ok = RTSC_DISABLED; |
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94 int len; |
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95 void *silence = NULL; |
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96 struct timeval start, end; |
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97 struct timespec delay; |
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98 int usec_delay; |
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99 unsigned last_samplecnt; |
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100 unsigned increment; |
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101 unsigned min_increment; |
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102 |
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103 len = 44100 * 4 / 4; /* amount of data for 0.25sec of 44.1khz, stereo, |
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104 * 16bit. 44kbyte can be sent to all supported |
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105 * sun audio devices without blocking in the |
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106 * "write" below. |
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107 */ |
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108 silence = calloc(1, len); |
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109 if (silence == NULL) |
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110 goto error; |
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111 |
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112 if ((fd = open(dev, O_WRONLY)) < 0) |
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113 goto error; |
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114 |
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115 AUDIO_INITINFO(&info); |
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116 info.play.sample_rate = 44100; |
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117 info.play.channels = AUDIO_CHANNELS_STEREO; |
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118 info.play.precision = AUDIO_PRECISION_16; |
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119 info.play.encoding = AUDIO_ENCODING_LINEAR; |
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120 info.play.samples = 0; |
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121 if (ioctl(fd, AUDIO_SETINFO, &info)) { |
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122 if (verbose) |
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123 printf("rtsc: SETINFO failed\n"); |
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124 goto error; |
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125 } |
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126 |
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127 if (write(fd, silence, len) != len) { |
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128 if (verbose) |
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129 printf("rtsc: write failed"); |
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130 goto error; |
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131 } |
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132 |
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133 if (ioctl(fd, AUDIO_GETINFO, &info)) { |
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134 if (verbose) |
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135 perror("rtsc: GETINFO1"); |
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136 goto error; |
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137 } |
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138 |
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139 last_samplecnt = info.play.samples; |
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140 min_increment = ~0; |
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141 |
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142 gettimeofday(&start, NULL); |
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143 for (;;) { |
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144 delay.tv_sec = 0; |
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145 delay.tv_nsec = 10000000; |
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146 nanosleep(&delay, NULL); |
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147 gettimeofday(&end, NULL); |
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148 usec_delay = (end.tv_sec - start.tv_sec) * 1000000 |
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149 + end.tv_usec - start.tv_usec; |
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150 |
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151 // stop monitoring sample counter after 0.2 seconds |
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152 if (usec_delay > 200000) |
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153 break; |
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154 |
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Applied patch by J¸«ärgen Keil (jk@tools.de), improves smoothness of video playback.
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155 if (ioctl(fd, AUDIO_GETINFO, &info)) { |
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156 if (verbose) |
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157 perror("rtsc: GETINFO2 failed"); |
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158 goto error; |
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159 } |
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160 if (info.play.samples < last_samplecnt) { |
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161 if (verbose) |
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162 printf("rtsc: %d > %d?\n", last_samplecnt, info.play.samples); |
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163 goto error; |
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164 } |
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165 |
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166 if ((increment = info.play.samples - last_samplecnt) > 0) { |
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167 if (verbose) |
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168 printf("ao_sun: sample counter increment: %d\n", increment); |
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169 if (increment < min_increment) { |
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170 min_increment = increment; |
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171 if (min_increment < 2000) |
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172 break; // looks good |
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173 } |
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174 } |
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175 last_samplecnt = info.play.samples; |
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176 } |
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177 |
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178 /* |
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179 * For 44.1kkz, stereo, 16-bit format we would send sound data in 16kbytes |
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180 * chunks (== 4096 samples) to the audio device. If we see a minimum |
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181 * sample counter increment from the soundcard driver of less than |
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182 * 2000 samples, we assume that the driver provides a useable realtime |
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183 * sample counter in the AUDIO_INFO play.samples field. Timing based |
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184 * on sample counts should be much more accurate than counting whole |
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185 * 16kbyte chunks. |
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186 */ |
1196
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187 if (min_increment < 2000) |
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188 rtsc_ok = RTSC_ENABLED; |
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189 |
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190 if (verbose) |
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191 printf("ao_sun: minimum sample counter increment per 10msec interval: %d\n" |
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192 "\t%susing sample counter based timing code\n", |
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193 min_increment, rtsc_ok == RTSC_ENABLED ? "" : "not "); |
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194 |
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195 |
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196 error: |
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197 if (silence != NULL) free(silence); |
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198 if (fd >= 0) { |
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199 #ifdef __svr4__ |
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200 // remove the 0 bytes from the above measurement from the |
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201 // audio driver's STREAMS queue |
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202 ioctl(fd, I_FLUSH, FLUSHW); |
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203 #endif |
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204 //ioctl(fd, AUDIO_DRAIN, 0); |
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205 close(fd); |
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206 } |
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207 |
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208 return rtsc_ok; |
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209 } |
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210 |
1041 | 211 // to set/get/query special features/parameters |
212 static int control(int cmd,int arg){ | |
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213 switch(cmd){ |
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214 case AOCONTROL_SET_DEVICE: |
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215 audio_dev=(char*)arg; |
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216 return CONTROL_OK; |
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217 case AOCONTROL_QUERY_FORMAT: |
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218 return CONTROL_TRUE; |
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219 case AOCONTROL_GET_VOLUME: |
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220 { |
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221 int fd,v,cmd,devs; |
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222 |
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223 fd=open( sun_mixer_device,O_RDONLY ); |
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224 if ( fd != -1 ) |
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225 { |
4907 | 226 ao_control_vol_t *vol = (ao_control_vol_t *)arg; |
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227 struct audio_info info; |
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228 ioctl( fd,AUDIO_GETINFO,&info); |
4907 | 229 vol->left=info.play.gain * 100. / AUDIO_MAX_GAIN; |
230 vol->right=info.play.gain * 100. / AUDIO_MAX_GAIN; | |
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231 close( fd ); |
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232 return CONTROL_OK; |
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233 } |
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234 return CONTROL_ERROR; |
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235 } |
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236 case AOCONTROL_SET_VOLUME: |
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237 { |
4907 | 238 ao_control_vol_t *vol = (ao_control_vol_t *)arg; |
4788
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239 int fd,v,cmd,devs; |
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240 |
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241 fd=open( sun_mixer_device,O_RDONLY ); |
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242 if ( fd != -1 ) |
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243 { |
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244 struct audio_info info; |
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245 AUDIO_INITINFO(&info); |
4907 | 246 info.play.gain = (vol->right+vol->left) * AUDIO_MAX_GAIN / 100 / 2; |
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247 ioctl( fd,AUDIO_SETINFO,&info ); |
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248 close( fd ); |
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249 return CONTROL_OK; |
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250 } |
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251 return CONTROL_ERROR; |
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252 } |
4907 | 253 } |
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254 return CONTROL_UNKNOWN; |
1041 | 255 } |
256 | |
257 // open & setup audio device | |
258 // return: 1=success 0=fail | |
259 static int init(int rate,int channels,int format,int flags){ | |
260 | |
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261 audio_info_t info; |
1807
6da3b9428ff0
Sun audiocs HW does not support AFMT_U8, try to use AFMT_S8 instead. If AFMT_S8
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|
262 int ok; |
1041 | 263 |
3492
24b0fad7fccc
Use $AUDIODEV environment variable as the default audio device on solaris, if
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diff
changeset
|
264 if (audio_dev == NULL) { |
24b0fad7fccc
Use $AUDIODEV environment variable as the default audio device on solaris, if
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265 if ((audio_dev = getenv("AUDIODEV")) == NULL) |
24b0fad7fccc
Use $AUDIODEV environment variable as the default audio device on solaris, if
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266 audio_dev = "/dev/audio"; |
24b0fad7fccc
Use $AUDIODEV environment variable as the default audio device on solaris, if
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|
267 } |
24b0fad7fccc
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|
268 |
5893
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|
269 if (sun_mixer_device == NULL) { |
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270 if ((sun_mixer_device = mixer_device) == NULL) { |
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271 sun_mixer_device = malloc(strlen(audio_dev) + 4); |
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272 strcpy(sun_mixer_device, audio_dev); |
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273 strcat(sun_mixer_device, "ctl"); |
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274 } |
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275 } |
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276 |
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277 if (ao_subdevice) audio_dev = ao_subdevice; |
1209 | 278 |
1308
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279 if (enable_sample_timing == RTSC_UNKNOWN |
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280 && !getenv("AO_SUN_DISABLE_SAMPLE_TIMING")) { |
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281 enable_sample_timing = realtime_samplecounter_available(audio_dev); |
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|
282 } |
1041 | 283 |
1456
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printfs cleanup - moved to higher -v level or moved to stderr
arpi
parents:
1346
diff
changeset
|
284 // printf("ao2: %d Hz %d chans %s [0x%X]\n", |
8c57a5a3c645
printfs cleanup - moved to higher -v level or moved to stderr
arpi
parents:
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|
285 // rate,channels,audio_out_format_name(format),format); |
1196
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Applied patch by J¸«ärgen Keil (jk@tools.de), improves smoothness of video playback.
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|
286 |
1308
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|
287 audio_fd=open(audio_dev, O_WRONLY); |
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288 if(audio_fd<0){ |
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289 printf("Can't open audio device %s, %s -> nosound\n", audio_dev, strerror(errno)); |
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290 return 0; |
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291 } |
1041 | 292 |
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293 ioctl(audio_fd, AUDIO_DRAIN, 0); |
1041 | 294 |
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295 AUDIO_INITINFO(&info); |
3095 | 296 info.play.encoding = oss2sunfmt(ao_data.format = format); |
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297 info.play.precision = |
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298 (format==AFMT_S16_LE || format==AFMT_S16_BE |
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299 ? AUDIO_PRECISION_16 |
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300 : AUDIO_PRECISION_8); |
3095 | 301 info.play.channels = ao_data.channels = channels; |
302 info.play.sample_rate = ao_data.samplerate = rate; | |
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303 convert_u8_s8 = 0; |
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304 ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0; |
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305 if (!ok && info.play.encoding == AUDIO_ENCODING_LINEAR8) { |
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306 /* sun audiocs hardware does not support U8 format, try S8... */ |
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307 info.play.encoding = AUDIO_ENCODING_LINEAR; |
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308 ok = ioctl(audio_fd, AUDIO_SETINFO, &info) >= 0; |
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309 if (ok) { |
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310 /* we must perform software U8 -> S8 conversion */ |
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311 convert_u8_s8 = 1; |
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312 } |
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313 } |
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314 if (!ok) { |
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315 printf("audio_setup: your card doesn't support %d channel, %s, %d Hz samplerate\n", |
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316 channels, audio_out_format_name(format), rate); |
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317 return 0; |
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318 } |
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319 |
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320 bytes_per_sample = channels * info.play.precision / 8; |
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321 byte_per_sec = bytes_per_sample * rate; |
3095 | 322 ao_data.outburst = byte_per_sec > 100000 ? 16384 : 8192; |
1041 | 323 |
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324 #ifdef __not_used__ |
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325 /* |
3095 | 326 * hmm, ao_data.buffersize is currently not used in this driver, do there's |
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327 * no need to measure it |
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328 */ |
3095 | 329 if(ao_data.buffersize==-1){ |
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330 // Measuring buffer size: |
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331 void* data; |
3095 | 332 ao_data.buffersize=0; |
1041 | 333 #ifdef HAVE_AUDIO_SELECT |
3095 | 334 data = malloc(ao_data.outburst); |
335 memset(data, format==AFMT_U8 ? 0x80 : 0, ao_data.outburst); | |
336 while(ao_data.buffersize<0x40000){ | |
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337 fd_set rfds; |
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338 struct timeval tv; |
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339 FD_ZERO(&rfds); FD_SET(audio_fd,&rfds); |
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340 tv.tv_sec=0; tv.tv_usec = 0; |
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341 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break; |
3095 | 342 write(audio_fd,data,ao_data.outburst); |
343 ao_data.buffersize+=ao_data.outburst; | |
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344 } |
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345 free(data); |
3095 | 346 if(ao_data.buffersize==0){ |
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347 printf("\n *** Your audio driver DOES NOT support select() ***\n"); |
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348 printf("Recompile mplayer with #undef HAVE_AUDIO_SELECT in config.h !\n\n"); |
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349 return 0; |
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350 } |
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351 #ifdef __svr4__ |
3095 | 352 // remove the 0 bytes from the above ao_data.buffersize measurement from the |
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353 // audio driver's STREAMS queue |
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354 ioctl(audio_fd, I_FLUSH, FLUSHW); |
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355 #endif |
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356 ioctl(audio_fd, AUDIO_DRAIN, 0); |
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357 #endif |
1041 | 358 } |
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359 #endif /* __not_used__ */ |
1041 | 360 |
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361 AUDIO_INITINFO(&info); |
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362 info.play.samples = 0; |
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363 info.play.eof = 0; |
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364 info.play.error = 0; |
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365 ioctl (audio_fd, AUDIO_SETINFO, &info); |
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366 |
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367 queued_bursts = 0; |
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368 queued_samples = 0; |
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369 |
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370 return 1; |
1041 | 371 } |
372 | |
373 // close audio device | |
374 static void uninit(){ | |
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375 #ifdef __svr4__ |
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376 // throw away buffered data in the audio driver's STREAMS queue |
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377 ioctl(audio_fd, I_FLUSH, FLUSHW); |
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378 #endif |
1041 | 379 close(audio_fd); |
380 } | |
381 | |
382 // stop playing and empty buffers (for seeking/pause) | |
383 static void reset(){ | |
384 audio_info_t info; | |
385 | |
386 uninit(); | |
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387 audio_fd=open(audio_dev, O_WRONLY); |
1041 | 388 if(audio_fd<0){ |
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389 printf("\nFatal error: *** CANNOT RE-OPEN / RESET AUDIO DEVICE (%s) ***\n", strerror(errno)); |
1041 | 390 return; |
391 } | |
392 | |
393 ioctl(audio_fd, AUDIO_DRAIN, 0); | |
394 | |
395 AUDIO_INITINFO(&info); | |
3095 | 396 info.play.encoding = oss2sunfmt(ao_data.format); |
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397 info.play.precision = |
3095 | 398 (ao_data.format==AFMT_S16_LE || ao_data.format==AFMT_S16_BE |
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399 ? AUDIO_PRECISION_16 |
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400 : AUDIO_PRECISION_8); |
3095 | 401 info.play.channels = ao_data.channels; |
402 info.play.sample_rate = ao_data.samplerate; | |
1041 | 403 info.play.samples = 0; |
404 info.play.eof = 0; | |
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405 info.play.error = 0; |
1041 | 406 ioctl (audio_fd, AUDIO_SETINFO, &info); |
407 queued_bursts = 0; | |
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408 queued_samples = 0; |
1041 | 409 } |
410 | |
411 // stop playing, keep buffers (for pause) | |
412 static void audio_pause() | |
413 { | |
414 struct audio_info info; | |
415 AUDIO_INITINFO(&info); | |
416 info.play.pause = 1; | |
417 ioctl(audio_fd, AUDIO_SETINFO, &info); | |
418 } | |
419 | |
420 // resume playing, after audio_pause() | |
421 static void audio_resume() | |
422 { | |
423 struct audio_info info; | |
424 AUDIO_INITINFO(&info); | |
425 info.play.pause = 0; | |
426 ioctl(audio_fd, AUDIO_SETINFO, &info); | |
427 } | |
428 | |
429 | |
430 // return: how many bytes can be played without blocking | |
431 static int get_space(){ | |
3095 | 432 int playsize = ao_data.outburst; |
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433 audio_info_t info; |
1041 | 434 |
435 // check buffer | |
436 #ifdef HAVE_AUDIO_SELECT | |
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437 { |
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438 fd_set rfds; |
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439 struct timeval tv; |
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440 FD_ZERO(&rfds); |
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441 FD_SET(audio_fd, &rfds); |
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442 tv.tv_sec = 0; |
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443 tv.tv_usec = 0; |
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444 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block! |
1041 | 445 } |
446 #endif | |
447 | |
5872 | 448 #ifndef __OpenBSD__ |
1041 | 449 ioctl(audio_fd, AUDIO_GETINFO, &info); |
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450 if (queued_bursts - info.play.eof > 2) |
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451 return 0; |
5872 | 452 #endif |
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453 |
3095 | 454 return ao_data.outburst; |
1041 | 455 } |
456 | |
457 // plays 'len' bytes of 'data' | |
458 // it should round it down to outburst*n | |
459 // return: number of bytes played | |
460 static int play(void* data,int len,int flags){ | |
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461 #if WORDS_BIGENDIAN |
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462 int native_endian = AFMT_S16_BE; |
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463 #else |
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464 int native_endian = AFMT_S16_LE; |
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465 #endif |
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466 |
3095 | 467 if (len < ao_data.outburst) return 0; |
468 len /= ao_data.outburst; | |
469 len *= ao_data.outburst; | |
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470 |
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471 /* 16-bit format using the 'wrong' byteorder? swap words */ |
3095 | 472 if ((ao_data.format == AFMT_S16_LE || ao_data.format == AFMT_S16_BE) |
473 && ao_data.format != native_endian) { | |
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474 static void *swab_buf; |
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475 static int swab_len; |
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476 if (len > swab_len) { |
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477 if (swab_buf) |
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478 swab_buf = realloc(swab_buf, len); |
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479 else |
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480 swab_buf = malloc(len); |
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481 swab_len = len; |
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482 if (swab_buf == NULL) return 0; |
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483 } |
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484 swab(data, swab_buf, len); |
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485 data = swab_buf; |
3095 | 486 } else if (ao_data.format == AFMT_U8 && convert_u8_s8) { |
1807
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487 int i; |
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488 unsigned char *p = data; |
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489 |
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490 for (i = 0, p = data; i < len; i++, p++) |
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491 *p ^= 0x80; |
1308
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492 } |
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493 |
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494 len = write(audio_fd, data, len); |
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495 if(len > 0) { |
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496 queued_samples += len / bytes_per_sample; |
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497 if (write(audio_fd,data,0) < 0) |
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498 perror("ao_sun: send EOF audio record"); |
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499 else |
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500 queued_bursts ++; |
1041 | 501 } |
502 return len; | |
503 } | |
504 | |
505 | |
3095 | 506 // return: delay in seconds between first and last sample in buffer |
507 static float get_delay(){ | |
1041 | 508 audio_info_t info; |
509 ioctl(audio_fd, AUDIO_GETINFO, &info); | |
5872 | 510 #ifdef __OpenBSD__ |
511 return (float) info.play.seek/ (float)byte_per_sec ; | |
512 #else | |
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513 if (info.play.samples && enable_sample_timing == RTSC_ENABLED) |
3095 | 514 return (float)(queued_samples - info.play.samples) / (float)byte_per_sec; |
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515 else |
3137 | 516 return (float)((queued_bursts - info.play.eof) * ao_data.outburst) / (float)byte_per_sec; |
5872 | 517 #endif |
1041 | 518 } |
519 |