Mercurial > mplayer.hg
annotate libao2/ao_sgi.c @ 33259:04dc3e55cd90
Increase the maximum value of the DVB timeout to 240 seconds.
Some devices may need more time for the initial tune (e.g. firmware loading).
Let the user specify higher timeout value if there is need to.
The default remains 30 seconds.
author | iive |
---|---|
date | Sun, 01 May 2011 18:07:59 +0000 |
parents | 32725ca88fed |
children |
rev | line source |
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2451 | 1 /* |
28343 | 2 * SGI/IRIX audio output driver |
3 * | |
4 * copyright (c) 2001 oliver.schoenbrunner@jku.at | |
5 * | |
6 * This file is part of MPlayer. | |
7 * | |
8 * MPlayer is free software; you can redistribute it and/or modify | |
9 * it under the terms of the GNU General Public License as published by | |
10 * the Free Software Foundation; either version 2 of the License, or | |
11 * (at your option) any later version. | |
12 * | |
13 * MPlayer is distributed in the hope that it will be useful, | |
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the | |
16 * GNU General Public License for more details. | |
17 * | |
18 * You should have received a copy of the GNU General Public License along | |
19 * with MPlayer; if not, write to the Free Software Foundation, Inc., | |
20 * 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA. | |
21 */ | |
2451 | 22 |
23 #include <stdio.h> | |
24 #include <stdlib.h> | |
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25 #include <unistd.h> |
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26 #include <errno.h> |
2451 | 27 #include <dmedia/audio.h> |
28 | |
29 #include "audio_out.h" | |
30 #include "audio_out_internal.h" | |
14123 | 31 #include "mp_msg.h" |
32 #include "help_mp.h" | |
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33 #include "libaf/af_format.h" |
2451 | 34 |
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35 static const ao_info_t info = |
2451 | 36 { |
37 "sgi audio output", | |
38 "sgi", | |
13549 | 39 "Oliver Schoenbrunner", |
2451 | 40 "" |
41 }; | |
42 | |
43 LIBAO_EXTERN(sgi) | |
44 | |
45 | |
46 static ALconfig ao_config; | |
47 static ALport ao_port; | |
11323 | 48 static int sample_rate; |
49 static int queue_size; | |
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50 static int bytes_per_frame; |
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51 |
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52 /** |
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53 * \param [in/out] format |
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54 * \param [out] width |
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55 * |
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56 * \return the closest matching SGI AL sample format |
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57 * |
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58 * \note width is set to required per-channel sample width |
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59 * format is updated to match the SGI AL sample format |
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60 */ |
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61 static int fmt2sgial(int *format, int *width) { |
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62 int smpfmt = AL_SAMPFMT_TWOSCOMP; |
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63 |
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64 /* SGI AL only supports float and signed integers in native |
24595 | 65 * endianness. If this is something else, we must rely on the audio |
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66 * filter to convert it to a compatible format. */ |
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67 |
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68 /* 24-bit audio is supported, but only with 32-bit alignment. |
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69 * mplayer's 24-bit format is packed, unfortunately. |
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70 * So we must upgrade 24-bit requests to 32 bits. Then we drop the |
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71 * lowest 8 bits during playback. */ |
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72 |
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73 switch(*format) { |
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74 case AF_FORMAT_U8: |
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75 case AF_FORMAT_S8: |
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76 *width = AL_SAMPLE_8; |
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77 *format = AF_FORMAT_S8; |
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78 break; |
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79 |
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80 case AF_FORMAT_U16_LE: |
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81 case AF_FORMAT_U16_BE: |
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82 case AF_FORMAT_S16_LE: |
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83 case AF_FORMAT_S16_BE: |
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84 *width = AL_SAMPLE_16; |
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85 *format = AF_FORMAT_S16_NE; |
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86 break; |
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87 |
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88 case AF_FORMAT_U24_LE: |
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89 case AF_FORMAT_U24_BE: |
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90 case AF_FORMAT_S24_LE: |
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91 case AF_FORMAT_S24_BE: |
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92 case AF_FORMAT_U32_LE: |
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93 case AF_FORMAT_U32_BE: |
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94 case AF_FORMAT_S32_LE: |
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95 case AF_FORMAT_S32_BE: |
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96 *width = AL_SAMPLE_24; |
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97 *format = AF_FORMAT_S32_NE; |
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98 break; |
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99 |
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100 case AF_FORMAT_FLOAT_LE: |
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101 case AF_FORMAT_FLOAT_BE: |
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102 *width = 4; |
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103 *format = AF_FORMAT_FLOAT_NE; |
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104 smpfmt = AL_SAMPFMT_FLOAT; |
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105 break; |
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106 |
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107 default: |
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108 *width = AL_SAMPLE_16; |
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109 *format = AF_FORMAT_S16_NE; |
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110 break; |
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111 |
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112 } |
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113 |
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114 return smpfmt; |
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115 } |
2451 | 116 |
117 // to set/get/query special features/parameters | |
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118 static int control(int cmd, void *arg){ |
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119 |
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120 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_INFO); |
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121 |
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122 switch(cmd) { |
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123 case AOCONTROL_QUERY_FORMAT: |
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124 /* Do not reject any format: return the closest matching |
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125 * format if the request is not supported natively. */ |
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126 return CONTROL_TRUE; |
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127 } |
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128 |
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129 return CONTROL_UNKNOWN; |
2451 | 130 } |
131 | |
132 // open & setup audio device | |
133 // return: 1=success 0=fail | |
134 static int init(int rate, int channels, int format, int flags) { | |
14249 | 135 |
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136 int smpwidth, smpfmt; |
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137 int rv = AL_DEFAULT_OUTPUT; |
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138 |
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139 smpfmt = fmt2sgial(&format, &smpwidth); |
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140 |
14264 | 141 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_InitInfo, rate, (channels > 1) ? "Stereo" : "Mono", af_fmt2str_short(format)); |
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142 |
2451 | 143 { /* from /usr/share/src/dmedia/audio/setrate.c */ |
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144 |
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145 double frate, realrate; |
2451 | 146 ALpv x[2]; |
147 | |
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148 if(ao_subdevice) { |
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149 rv = alGetResourceByName(AL_SYSTEM, ao_subdevice, AL_OUTPUT_DEVICE_TYPE); |
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150 if (!rv) { |
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151 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InvalidDevice); |
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152 return 0; |
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153 } |
2451 | 154 } |
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155 |
2451 | 156 frate = rate; |
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157 |
2451 | 158 x[0].param = AL_RATE; |
159 x[0].value.ll = alDoubleToFixed(rate); | |
160 x[1].param = AL_MASTER_CLOCK; | |
161 x[1].value.i = AL_CRYSTAL_MCLK_TYPE; | |
162 | |
163 if (alSetParams(rv,x, 2)<0) { | |
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164 mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetParms_Samplerate, alGetErrorString(oserror())); |
2451 | 165 } |
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166 |
2451 | 167 if (x[0].sizeOut < 0) { |
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168 mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantSetAlRate); |
2451 | 169 } |
170 | |
171 if (alGetParams(rv,x, 1)<0) { | |
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172 mp_msg(MSGT_AO, MSGL_WARN, MSGTR_AO_SGI_CantGetParms, alGetErrorString(oserror())); |
2451 | 173 } |
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174 |
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175 realrate = alFixedToDouble(x[0].value.ll); |
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176 if (frate != realrate) { |
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177 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_SampleRateInfo, realrate, frate); |
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178 } |
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179 sample_rate = (int)realrate; |
2451 | 180 } |
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181 |
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182 bytes_per_frame = channels * smpwidth; |
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183 |
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184 ao_data.samplerate = sample_rate; |
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185 ao_data.channels = channels; |
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186 ao_data.format = format; |
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187 ao_data.bps = sample_rate * bytes_per_frame; |
3095 | 188 ao_data.buffersize=131072; |
189 ao_data.outburst = ao_data.buffersize/16; | |
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190 |
2451 | 191 ao_config = alNewConfig(); |
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192 |
2451 | 193 if (!ao_config) { |
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194 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror())); |
2451 | 195 return 0; |
196 } | |
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197 |
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198 if(alSetChannels(ao_config, channels) < 0 || |
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199 alSetWidth(ao_config, smpwidth) < 0 || |
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200 alSetSampFmt(ao_config, smpfmt) < 0 || |
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201 alSetQueueSize(ao_config, sample_rate) < 0 || |
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202 alSetDevice(ao_config, rv) < 0) { |
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203 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitConfigError, alGetErrorString(oserror())); |
2451 | 204 return 0; |
205 } | |
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206 |
2451 | 207 ao_port = alOpenPort("mplayer", "w", ao_config); |
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208 |
2451 | 209 if (!ao_port) { |
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210 mp_msg(MSGT_AO, MSGL_ERR, MSGTR_AO_SGI_InitOpenAudioFailed, alGetErrorString(oserror())); |
2451 | 211 return 0; |
212 } | |
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213 |
2451 | 214 // printf("ao_sgi, init: port %d config %d\n", ao_port, ao_config); |
11323 | 215 queue_size = alGetQueueSize(ao_config); |
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216 return 1; |
2451 | 217 |
218 } | |
219 | |
220 // close audio device | |
12145 | 221 static void uninit(int immed) { |
2451 | 222 |
223 /* TODO: samplerate should be set back to the value before mplayer was started! */ | |
224 | |
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225 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Uninit); |
2451 | 226 |
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227 if (ao_config) { |
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228 alFreeConfig(ao_config); |
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229 ao_config = NULL; |
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230 } |
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231 |
2451 | 232 if (ao_port) { |
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233 if (!immed) |
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234 while(alGetFilled(ao_port) > 0) sginap(1); |
2451 | 235 alClosePort(ao_port); |
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236 ao_port = NULL; |
2451 | 237 } |
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238 |
2451 | 239 } |
240 | |
241 // stop playing and empty buffers (for seeking/pause) | |
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242 static void reset(void) { |
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243 |
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244 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_Reset); |
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245 |
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246 alDiscardFrames(ao_port, queue_size); |
2451 | 247 } |
248 | |
249 // stop playing, keep buffers (for pause) | |
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250 static void audio_pause(void) { |
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251 |
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252 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_PauseInfo); |
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253 |
2451 | 254 } |
255 | |
256 // resume playing, after audio_pause() | |
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257 static void audio_resume(void) { |
2451 | 258 |
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259 mp_msg(MSGT_AO, MSGL_INFO, MSGTR_AO_SGI_ResumeInfo); |
2451 | 260 |
261 } | |
262 | |
263 // return: how many bytes can be played without blocking | |
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264 static int get_space(void) { |
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265 |
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266 // printf("ao_sgi, get_space: (ao_outburst %d)\n", ao_data.outburst); |
2451 | 267 // printf("ao_sgi, get_space: alGetFillable [%d] \n", alGetFillable(ao_port)); |
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268 |
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269 return alGetFillable(ao_port) * bytes_per_frame; |
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270 |
2451 | 271 } |
272 | |
273 | |
274 // plays 'len' bytes of 'data' | |
275 // it should round it down to outburst*n | |
276 // return: number of bytes played | |
277 static int play(void* data, int len, int flags) { | |
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278 |
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279 /* Always process data in quadword-aligned chunks (64-bits). */ |
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280 const int plen = len / (sizeof(uint64_t) * bytes_per_frame); |
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281 const int framecount = plen * sizeof(uint64_t); |
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282 |
2451 | 283 // printf("ao_sgi, play: len %d flags %d (%d %d)\n", len, flags, ao_port, ao_config); |
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284 // printf("channels %d\n", ao_data.channels); |
2451 | 285 |
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286 if(ao_data.format == AF_FORMAT_S32_NE) { |
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287 /* The zen of this is explained in fmt2sgial() */ |
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288 int32_t *smpls = data; |
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289 const int32_t *smple = smpls + (framecount * ao_data.channels); |
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290 while(smpls < smple) |
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291 *smpls++ >>= 8; |
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292 } |
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293 |
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294 alWriteFrames(ao_port, data, framecount); |
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295 |
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296 return framecount * bytes_per_frame; |
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297 |
2451 | 298 } |
299 | |
3095 | 300 // return: delay in seconds between first and last sample in buffer |
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301 static float get_delay(void){ |
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302 |
2451 | 303 // printf("ao_sgi, get_delay: (ao_buffersize %d)\n", ao_buffersize); |
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304 |
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305 // return (float)queue_size/((float)sample_rate); |
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306 const int outstanding = alGetFilled(ao_port); |
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307 return (float)((outstanding < 0) ? queue_size : outstanding) / |
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308 ((float)sample_rate); |
2451 | 309 } |