Mercurial > mplayer.hg
annotate libao2/ao_oss.c @ 15010:aec21cf429cb
Improved encoding guide:
- explains how to do a smart resize
- calculate a bitrate to target a certain size
- rip audio and transcode in Ogg/Vorbis
- a lot of libavcodec's options
to come: muxing in matroska files...
author | gpoirier |
---|---|
date | Fri, 25 Mar 2005 21:17:21 +0000 |
parents | 2d919650990a |
children | a45af202818f |
rev | line source |
---|---|
954 | 1 #include <stdio.h> |
2 #include <stdlib.h> | |
3 | |
4 #include <sys/ioctl.h> | |
5 #include <unistd.h> | |
6 #include <sys/time.h> | |
7 #include <sys/types.h> | |
8 #include <sys/stat.h> | |
9 #include <fcntl.h> | |
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10 #include <errno.h> |
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11 #include <string.h> |
954 | 12 |
14123 | 13 #include "config.h" |
14 #include "mp_msg.h" | |
15 #include "mixer.h" | |
16 #include "help_mp.h" | |
954 | 17 |
14245 | 18 #ifdef HAVE_SYS_SOUNDCARD_H |
19 #include <sys/soundcard.h> | |
20 #else | |
21 #ifdef HAVE_SOUNDCARD_H | |
22 #include <soundcard.h> | |
23 #endif | |
24 #endif | |
25 | |
26 #include "../libaf/af_format.h" | |
1532 | 27 |
954 | 28 #include "audio_out.h" |
29 #include "audio_out_internal.h" | |
30 | |
31 static ao_info_t info = | |
32 { | |
33 "OSS/ioctl audio output", | |
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34 "oss", |
954 | 35 "A'rpi", |
36 "" | |
37 }; | |
38 | |
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39 /* Support for >2 output channels added 2001-11-25 - Steve Davies <steve@daviesfam.org> */ |
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40 |
954 | 41 LIBAO_EXTERN(oss) |
42 | |
14245 | 43 static int format2oss(int format) |
44 { | |
45 switch(format) | |
46 { | |
47 case AF_FORMAT_U8: return AFMT_U8; | |
48 case AF_FORMAT_S8: return AFMT_S8; | |
49 case AF_FORMAT_U16_LE: return AFMT_U16_LE; | |
50 case AF_FORMAT_U16_BE: return AFMT_U16_BE; | |
51 case AF_FORMAT_S16_LE: return AFMT_S16_LE; | |
52 case AF_FORMAT_S16_BE: return AFMT_S16_BE; | |
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53 #ifdef AFMT_U24_LE |
14245 | 54 case AF_FORMAT_U24_LE: return AFMT_U24_LE; |
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55 #endif |
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56 #ifdef AFMT_U24_BE |
14245 | 57 case AF_FORMAT_U24_BE: return AFMT_U24_BE; |
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58 #endif |
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59 #ifdef AFMT_S24_LE |
14245 | 60 case AF_FORMAT_S24_LE: return AFMT_S24_LE; |
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61 #endif |
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62 #ifdef AFMT_S24_BE |
14245 | 63 case AF_FORMAT_S24_BE: return AFMT_S24_BE; |
64 #endif | |
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65 #ifdef AFMT_U32_LE |
14245 | 66 case AF_FORMAT_U32_LE: return AFMT_U32_LE; |
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67 #endif |
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68 #ifdef AFMT_U32_BE |
14245 | 69 case AF_FORMAT_U32_BE: return AFMT_U32_BE; |
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70 #endif |
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71 #ifdef AFMT_S32_LE |
14245 | 72 case AF_FORMAT_S32_LE: return AFMT_S32_LE; |
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73 #endif |
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74 #ifdef AFMT_S32_BE |
14245 | 75 case AF_FORMAT_S32_BE: return AFMT_S32_BE; |
76 #endif | |
77 #ifdef AFMT_FLOAT | |
78 case AF_FORMAT_FLOAT_NE: return AFMT_FLOAT; | |
79 #endif | |
80 // SPECIALS | |
81 case AF_FORMAT_MU_LAW: return AFMT_MU_LAW; | |
82 case AF_FORMAT_A_LAW: return AFMT_A_LAW; | |
83 case AF_FORMAT_IMA_ADPCM: return AFMT_IMA_ADPCM; | |
84 #ifdef AFMT_MPEG | |
85 case AF_FORMAT_MPEG2: return AFMT_MPEG; | |
86 #endif | |
87 #ifdef AFMT_AC3 | |
88 case AF_FORMAT_AC3: return AFMT_AC3; | |
89 #endif | |
90 } | |
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91 mp_msg(MSGT_AO, MSGL_V, "OSS: Unknown/not supported internal format: %s\n", af_fmt2str_short(format)); |
14245 | 92 return -1; |
93 } | |
94 | |
95 static int oss2format(int format) | |
96 { | |
97 switch(format) | |
98 { | |
99 case AFMT_U8: return AF_FORMAT_U8; | |
100 case AFMT_S8: return AF_FORMAT_S8; | |
101 case AFMT_U16_LE: return AF_FORMAT_U16_LE; | |
102 case AFMT_U16_BE: return AF_FORMAT_U16_BE; | |
103 case AFMT_S16_LE: return AF_FORMAT_S16_LE; | |
104 case AFMT_S16_BE: return AF_FORMAT_S16_BE; | |
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105 #ifdef AFMT_U24_LE |
14245 | 106 case AFMT_U24_LE: return AF_FORMAT_U24_LE; |
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107 #endif |
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108 #ifdef AFMT_U24_BE |
14245 | 109 case AFMT_U24_BE: return AF_FORMAT_U24_BE; |
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110 #endif |
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111 #ifdef AFMT_S24_LE |
14245 | 112 case AFMT_S24_LE: return AF_FORMAT_S24_LE; |
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113 #endif |
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114 #ifdef AFMT_S24_BE |
14245 | 115 case AFMT_S24_BE: return AF_FORMAT_S24_BE; |
116 #endif | |
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117 #ifdef AFMT_U32_LE |
14245 | 118 case AFMT_U32_LE: return AF_FORMAT_U32_LE; |
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119 #endif |
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120 #ifdef AFMT_U32_BE |
14245 | 121 case AFMT_U32_BE: return AF_FORMAT_U32_BE; |
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122 #endif |
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123 #ifdef AFMT_S32_LE |
14245 | 124 case AFMT_S32_LE: return AF_FORMAT_S32_LE; |
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125 #endif |
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126 #ifdef AFMT_S32_BE |
14245 | 127 case AFMT_S32_BE: return AF_FORMAT_S32_BE; |
128 #endif | |
129 #ifdef AFMT_FLOAT | |
130 case AFMT_FLOAT: return AF_FORMAT_FLOAT_NE; | |
131 #endif | |
132 // SPECIALS | |
133 case AFMT_MU_LAW: return AF_FORMAT_MU_LAW; | |
134 case AFMT_A_LAW: return AF_FORMAT_A_LAW; | |
135 case AFMT_IMA_ADPCM: return AF_FORMAT_IMA_ADPCM; | |
136 #ifdef AFMT_MPEG | |
137 case AFMT_MPEG: return AF_FORMAT_MPEG2; | |
138 #endif | |
139 #ifdef AFMT_AC3 | |
140 case AFMT_AC3: return AF_FORMAT_AC3; | |
141 #endif | |
142 } | |
14273 | 143 printf("Unknown/not supported OSS format: %x\n", format); |
14245 | 144 return -1; |
145 } | |
146 | |
4803 | 147 static char *dsp=PATH_DEV_DSP; |
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148 static audio_buf_info zz; |
954 | 149 static int audio_fd=-1; |
150 | |
4803 | 151 char *oss_mixer_device = PATH_DEV_MIXER; |
11837 | 152 int oss_mixer_channel = SOUND_MIXER_PCM; |
1191 | 153 |
954 | 154 // to set/get/query special features/parameters |
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155 static int control(int cmd,void *arg){ |
954 | 156 switch(cmd){ |
157 case AOCONTROL_SET_DEVICE: | |
158 dsp=(char*)arg; | |
159 return CONTROL_OK; | |
6795 | 160 case AOCONTROL_GET_DEVICE: |
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161 *(char**)arg=dsp; |
6795 | 162 return CONTROL_OK; |
954 | 163 case AOCONTROL_QUERY_FORMAT: |
164 return CONTROL_TRUE; | |
1191 | 165 case AOCONTROL_GET_VOLUME: |
166 case AOCONTROL_SET_VOLUME: | |
167 { | |
168 ao_control_vol_t *vol = (ao_control_vol_t *)arg; | |
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169 int fd, v, devs; |
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170 |
14245 | 171 if(ao_data.format == AF_FORMAT_AC3) |
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172 return CONTROL_TRUE; |
1191 | 173 |
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174 if ((fd = open(oss_mixer_device, O_RDONLY)) > 0) |
1191 | 175 { |
176 ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); | |
11837 | 177 if (devs & (1 << oss_mixer_channel)) |
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178 { |
1191 | 179 if (cmd == AOCONTROL_GET_VOLUME) |
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180 { |
11837 | 181 ioctl(fd, MIXER_READ(oss_mixer_channel), &v); |
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182 vol->right = (v & 0xFF00) >> 8; |
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183 vol->left = v & 0x00FF; |
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184 } |
1191 | 185 else |
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186 { |
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187 v = ((int)vol->right << 8) | (int)vol->left; |
11837 | 188 ioctl(fd, MIXER_WRITE(oss_mixer_channel), &v); |
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189 } |
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190 } |
1191 | 191 else |
192 { | |
193 close(fd); | |
194 return CONTROL_ERROR; | |
195 } | |
196 close(fd); | |
197 return CONTROL_OK; | |
198 } | |
199 } | |
200 return CONTROL_ERROR; | |
954 | 201 } |
202 return CONTROL_UNKNOWN; | |
203 } | |
204 | |
205 // open & setup audio device | |
206 // return: 1=success 0=fail | |
207 static int init(int rate,int channels,int format,int flags){ | |
11837 | 208 char *mixer_channels [SOUND_MIXER_NRDEVICES] = SOUND_DEVICE_NAMES; |
14245 | 209 int oss_format; |
954 | 210 |
14264 | 211 mp_msg(MSGT_AO,MSGL_V,"ao2: %d Hz %d chans %s\n",rate,channels, |
212 af_fmt2str_short(format)); | |
954 | 213 |
1191 | 214 if (ao_subdevice) |
215 dsp = ao_subdevice; | |
216 | |
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217 if(mixer_device) |
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218 oss_mixer_device=mixer_device; |
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219 |
11837 | 220 if(mixer_channel){ |
221 int fd, devs, i; | |
222 | |
223 if ((fd = open(oss_mixer_device, O_RDONLY)) == -1){ | |
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224 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenMixer, |
11837 | 225 oss_mixer_device, strerror(errno)); |
226 }else{ | |
227 ioctl(fd, SOUND_MIXER_READ_DEVMASK, &devs); | |
228 close(fd); | |
229 | |
230 for (i=0; i<SOUND_MIXER_NRDEVICES; i++){ | |
231 if(!strcasecmp(mixer_channels[i], mixer_channel)){ | |
232 if(!(devs & (1 << i))){ | |
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233 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound, |
11837 | 234 mixer_channel); |
235 i = SOUND_MIXER_NRDEVICES+1; | |
236 break; | |
237 } | |
238 oss_mixer_channel = i; | |
239 break; | |
240 } | |
241 } | |
242 if(i==SOUND_MIXER_NRDEVICES){ | |
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243 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_ChanNotFound, |
11837 | 244 mixer_channel); |
245 } | |
246 } | |
247 } | |
248 | |
6296 | 249 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' dsp device\n", dsp); |
11837 | 250 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", oss_mixer_device); |
251 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using '%s' mixer device\n", mixer_channels[oss_mixer_channel]); | |
1191 | 252 |
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253 #ifdef __linux__ |
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254 audio_fd=open(dsp, O_WRONLY | O_NONBLOCK); |
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255 #else |
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256 audio_fd=open(dsp, O_WRONLY); |
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257 #endif |
954 | 258 if(audio_fd<0){ |
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259 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantOpenDev, dsp, strerror(errno)); |
954 | 260 return 0; |
261 } | |
262 | |
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263 #ifdef __linux__ |
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264 /* Remove the non-blocking flag */ |
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265 if(fcntl(audio_fd, F_SETFL, 0) < 0) { |
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266 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantMakeFd, strerror(errno)); |
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267 return 0; |
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268 } |
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269 #endif |
8303 | 270 |
271 #if defined(FD_CLOEXEC) && defined(F_SETFD) | |
272 fcntl(audio_fd, F_SETFD, FD_CLOEXEC); | |
273 #endif | |
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274 |
14245 | 275 if(format == AF_FORMAT_AC3) { |
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276 ao_data.samplerate=rate; |
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277 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); |
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278 } |
5790 | 279 |
280 ac3_retry: | |
3095 | 281 ao_data.format=format; |
14245 | 282 oss_format=format2oss(format); |
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283 if (oss_format == -1) { |
5790 | 284 #ifdef WORDS_BIGENDIAN |
14245 | 285 oss_format=AFMT_S16_BE; |
5790 | 286 #else |
14245 | 287 oss_format=AFMT_S16_LE; |
5790 | 288 #endif |
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289 format=AF_FORMAT_S16_NE; |
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290 } |
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291 if( ioctl(audio_fd, SNDCTL_DSP_SETFMT, &oss_format)<0 || |
14834
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Have OSS audio out fall back to s16ne instead of u8 if it can't open the
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292 oss_format != format2oss(format)) { |
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Have OSS audio out fall back to s16ne instead of u8 if it can't open the
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293 mp_msg(MSGT_AO,MSGL_WARN, MSGTR_AO_OSS_CantSet, dsp, |
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294 af_fmt2str_short(format), af_fmt2str_short(AF_FORMAT_S16_NE) ); |
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295 format=AF_FORMAT_S16_NE; |
5790 | 296 goto ac3_retry; |
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297 } |
7668 | 298 #if 0 |
14245 | 299 if(oss_format!=format2oss(format)) |
6899 | 300 mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %s sample format! Broken audio or bad playback speed are possible! Try with '-aop list=format'\n",audio_out_format_name(format)); |
7668 | 301 #endif |
14245 | 302 |
303 ao_data.format = oss2format(oss_format); | |
304 if (ao_data.format == -1) return 0; | |
14264 | 305 |
306 mp_msg(MSGT_AO,MSGL_V,"audio_setup: sample format: %s (requested: %s)\n", | |
307 af_fmt2str_short(ao_data.format), af_fmt2str_short(format)); | |
954 | 308 |
8848 | 309 ao_data.channels = channels; |
14245 | 310 if(format != AF_FORMAT_AC3) { |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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311 // We only use SNDCTL_DSP_CHANNELS for >2 channels, in case some drivers don't have it |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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312 if (ao_data.channels > 2) { |
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fixed to check that SNDCTL_DSP_CHANNELS actually grants the requested number of channels
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313 if ( ioctl(audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels) == -1 || |
66134af21278
fixed to check that SNDCTL_DSP_CHANNELS actually grants the requested number of channels
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314 ao_data.channels != channels ) { |
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315 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, channels); |
3181
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316 return 0; |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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317 } |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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318 } |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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319 else { |
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320 int c = ao_data.channels-1; |
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321 if (ioctl (audio_fd, SNDCTL_DSP_STEREO, &c) == -1) { |
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322 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantSetChans, ao_data.channels); |
3181
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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323 return 0; |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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324 } |
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nice 10l bug, found by Balatoni Denes <pnis@coder.hu>
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325 ao_data.channels=c+1; |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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326 } |
6296 | 327 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d channels (requested: %d)\n", ao_data.channels, channels); |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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328 // set rate |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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329 ao_data.samplerate=rate; |
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Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
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330 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); |
6296 | 331 mp_msg(MSGT_AO,MSGL_V,"audio_setup: using %d Hz samplerate (requested: %d)\n",ao_data.samplerate,rate); |
7668 | 332 #if 0 |
6296 | 333 if(ao_data.samplerate!=rate) |
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334 mp_msg(MSGT_AO,MSGL_WARN,"WARNING! Your soundcard does NOT support %d Hz samplerate! A-V sync problems or wrong speed are possible! Try with '-aop list=resample:fout=%d'\n",rate,ao_data.samplerate); |
7668 | 335 #endif |
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336 } |
954 | 337 |
338 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)==-1){ | |
339 int r=0; | |
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340 mp_msg(MSGT_AO,MSGL_WARN,MSGTR_AO_OSS_CantUseGetospace); |
954 | 341 if(ioctl(audio_fd, SNDCTL_DSP_GETBLKSIZE, &r)==-1){ |
6296 | 342 mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (config.h)\n",ao_data.outburst); |
954 | 343 } else { |
3095 | 344 ao_data.outburst=r; |
6296 | 345 mp_msg(MSGT_AO,MSGL_V,"audio_setup: %d bytes/frag (GETBLKSIZE)\n",ao_data.outburst); |
954 | 346 } |
347 } else { | |
6296 | 348 mp_msg(MSGT_AO,MSGL_V,"audio_setup: frags: %3d/%d (%d bytes/frag) free: %6d\n", |
954 | 349 zz.fragments, zz.fragstotal, zz.fragsize, zz.bytes); |
3095 | 350 if(ao_data.buffersize==-1) ao_data.buffersize=zz.bytes; |
351 ao_data.outburst=zz.fragsize; | |
954 | 352 } |
353 | |
3095 | 354 if(ao_data.buffersize==-1){ |
954 | 355 // Measuring buffer size: |
356 void* data; | |
3095 | 357 ao_data.buffersize=0; |
954 | 358 #ifdef HAVE_AUDIO_SELECT |
3095 | 359 data=malloc(ao_data.outburst); memset(data,0,ao_data.outburst); |
360 while(ao_data.buffersize<0x40000){ | |
954 | 361 fd_set rfds; |
362 struct timeval tv; | |
363 FD_ZERO(&rfds); FD_SET(audio_fd,&rfds); | |
364 tv.tv_sec=0; tv.tv_usec = 0; | |
365 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) break; | |
3095 | 366 write(audio_fd,data,ao_data.outburst); |
367 ao_data.buffersize+=ao_data.outburst; | |
954 | 368 } |
369 free(data); | |
3095 | 370 if(ao_data.buffersize==0){ |
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371 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantUseSelect); |
954 | 372 return 0; |
373 } | |
374 #endif | |
375 } | |
376 | |
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377 ao_data.bps=ao_data.channels; |
14245 | 378 if(ao_data.format != AF_FORMAT_U8 && ao_data.format != AF_FORMAT_S8) |
8603
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379 ao_data.bps*=2; |
30bef3c97b8b
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380 |
6075
0b3b6d03779c
fix for L<->R swap with 6ch playback - based on patch by Eric Lammerts <eric@lammerts.org>
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381 ao_data.outburst-=ao_data.outburst % ao_data.bps; // round down |
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382 ao_data.bps*=ao_data.samplerate; |
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383 |
954 | 384 return 1; |
385 } | |
386 | |
387 // close audio device | |
12145 | 388 static void uninit(int immed){ |
7118
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389 if(audio_fd == -1) return; |
12145 | 390 #ifdef SNDCTL_DSP_SYNC |
391 // to get the buffer played | |
392 if (!immed) | |
393 ioctl(audio_fd, SNDCTL_DSP_SYNC, NULL); | |
394 #endif | |
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395 #ifdef SNDCTL_DSP_RESET |
12145 | 396 if (immed) |
397 ioctl(audio_fd, SNDCTL_DSP_RESET, NULL); | |
1020
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398 #endif |
954 | 399 close(audio_fd); |
7118
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changeset
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400 audio_fd = -1; |
954 | 401 } |
402 | |
403 // stop playing and empty buffers (for seeking/pause) | |
404 static void reset(){ | |
14245 | 405 int oss_format; |
12145 | 406 uninit(1); |
5907
d8c34a35ccb9
nonblock open is not legal for OSS (see OSS api docs), and causes problems on freebsd (and maybe other OSs)
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changeset
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407 audio_fd=open(dsp, O_WRONLY); |
d8c34a35ccb9
nonblock open is not legal for OSS (see OSS api docs), and causes problems on freebsd (and maybe other OSs)
arpi
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408 if(audio_fd < 0){ |
13383
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ivo
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|
409 mp_msg(MSGT_AO,MSGL_ERR,MSGTR_AO_OSS_CantReopen, strerror(errno)); |
954 | 410 return; |
411 } | |
412 | |
10600
3cf428b56cde
Set close-on-exec after newer open (in reset func). Patch by Andriy Gapon <agapon@cv-nj.com>
alex
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|
413 #if defined(FD_CLOEXEC) && defined(F_SETFD) |
3cf428b56cde
Set close-on-exec after newer open (in reset func). Patch by Andriy Gapon <agapon@cv-nj.com>
alex
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changeset
|
414 fcntl(audio_fd, F_SETFD, FD_CLOEXEC); |
3cf428b56cde
Set close-on-exec after newer open (in reset func). Patch by Andriy Gapon <agapon@cv-nj.com>
alex
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415 #endif |
3cf428b56cde
Set close-on-exec after newer open (in reset func). Patch by Andriy Gapon <agapon@cv-nj.com>
alex
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changeset
|
416 |
14245 | 417 oss_format = format2oss(ao_data.format); |
418 ioctl (audio_fd, SNDCTL_DSP_SETFMT, &oss_format); | |
419 if(ao_data.format != AF_FORMAT_AC3) { | |
3181
c8edb0691f09
Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
steve
parents:
3137
diff
changeset
|
420 if (ao_data.channels > 2) |
c8edb0691f09
Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
steve
parents:
3137
diff
changeset
|
421 ioctl (audio_fd, SNDCTL_DSP_CHANNELS, &ao_data.channels); |
c8edb0691f09
Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
steve
parents:
3137
diff
changeset
|
422 else { |
c8edb0691f09
Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
steve
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diff
changeset
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423 int c = ao_data.channels-1; |
c8edb0691f09
Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
steve
parents:
3137
diff
changeset
|
424 ioctl (audio_fd, SNDCTL_DSP_STEREO, &c); |
c8edb0691f09
Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
steve
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changeset
|
425 } |
c8edb0691f09
Extended oss output driver and libac3 to support 4 and 6 channel output mixes. added -channels command line option
steve
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diff
changeset
|
426 ioctl (audio_fd, SNDCTL_DSP_SPEED, &ao_data.samplerate); |
1528
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ac3/spdif patch by German Gomez Garcia <german@piraos.com>
arpi
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|
427 } |
954 | 428 } |
429 | |
1038
b36fb1ae4b53
applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
arpi_esp
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1020
diff
changeset
|
430 // stop playing, keep buffers (for pause) |
b36fb1ae4b53
applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
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|
431 static void audio_pause() |
b36fb1ae4b53
applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
arpi_esp
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1020
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changeset
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432 { |
12145 | 433 uninit(1); |
1038
b36fb1ae4b53
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arpi_esp
parents:
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diff
changeset
|
434 } |
b36fb1ae4b53
applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
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diff
changeset
|
435 |
b36fb1ae4b53
applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
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parents:
1020
diff
changeset
|
436 // resume playing, after audio_pause() |
b36fb1ae4b53
applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
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diff
changeset
|
437 static void audio_resume() |
b36fb1ae4b53
applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
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diff
changeset
|
438 { |
7118
4332922afec4
release /dev/dsp on pause patch by Eric Lammerts <eric@lammerts.org> (and D Richard Felker III <dalias@aerifal.cx>)
alex
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diff
changeset
|
439 reset(); |
1038
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applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
arpi_esp
parents:
1020
diff
changeset
|
440 } |
b36fb1ae4b53
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arpi_esp
parents:
1020
diff
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|
441 |
b36fb1ae4b53
applied solaris8/netbsd/other fixes patch by J¸«ärgen Keil <jk@tools.de>
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parents:
1020
diff
changeset
|
442 |
954 | 443 // return: how many bytes can be played without blocking |
444 static int get_space(){ | |
3095 | 445 int playsize=ao_data.outburst; |
954 | 446 |
1020
72cacd3b8f30
Solaris 8 support - patch by Marcus Comstedt <marcus@idonex.se>
arpi_esp
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956
diff
changeset
|
447 #ifdef SNDCTL_DSP_GETOSPACE |
954 | 448 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1){ |
449 // calculate exact buffer space: | |
3455
921a78c7b4aa
limit get_space return <= MAX_OUTBURST, whilst always an exact number of fragments
steve
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3319
diff
changeset
|
450 playsize = zz.fragments*zz.fragsize; |
921a78c7b4aa
limit get_space return <= MAX_OUTBURST, whilst always an exact number of fragments
steve
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diff
changeset
|
451 if (playsize > MAX_OUTBURST) |
921a78c7b4aa
limit get_space return <= MAX_OUTBURST, whilst always an exact number of fragments
steve
parents:
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diff
changeset
|
452 playsize = (MAX_OUTBURST / zz.fragsize) * zz.fragsize; |
921a78c7b4aa
limit get_space return <= MAX_OUTBURST, whilst always an exact number of fragments
steve
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3319
diff
changeset
|
453 return playsize; |
954 | 454 } |
1020
72cacd3b8f30
Solaris 8 support - patch by Marcus Comstedt <marcus@idonex.se>
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diff
changeset
|
455 #endif |
954 | 456 |
457 // check buffer | |
458 #ifdef HAVE_AUDIO_SELECT | |
459 { fd_set rfds; | |
460 struct timeval tv; | |
461 FD_ZERO(&rfds); | |
462 FD_SET(audio_fd, &rfds); | |
463 tv.tv_sec = 0; | |
464 tv.tv_usec = 0; | |
465 if(!select(audio_fd+1, NULL, &rfds, NULL, &tv)) return 0; // not block! | |
466 } | |
467 #endif | |
468 | |
3095 | 469 return ao_data.outburst; |
954 | 470 } |
471 | |
472 // plays 'len' bytes of 'data' | |
473 // it should round it down to outburst*n | |
474 // return: number of bytes played | |
475 static int play(void* data,int len,int flags){ | |
3095 | 476 len/=ao_data.outburst; |
477 len=write(audio_fd,data,len*ao_data.outburst); | |
954 | 478 return len; |
479 } | |
480 | |
481 static int audio_delay_method=2; | |
482 | |
3095 | 483 // return: delay in seconds between first and last sample in buffer |
484 static float get_delay(){ | |
485 /* Calculate how many bytes/second is sent out */ | |
954 | 486 if(audio_delay_method==2){ |
5872 | 487 #ifdef SNDCTL_DSP_GETODELAY |
954 | 488 int r=0; |
489 if(ioctl(audio_fd, SNDCTL_DSP_GETODELAY, &r)!=-1) | |
3095 | 490 return ((float)r)/(float)ao_data.bps; |
5872 | 491 #endif |
954 | 492 audio_delay_method=1; // fallback if not supported |
493 } | |
494 if(audio_delay_method==1){ | |
495 // SNDCTL_DSP_GETOSPACE | |
496 if(ioctl(audio_fd, SNDCTL_DSP_GETOSPACE, &zz)!=-1) | |
3095 | 497 return ((float)(ao_data.buffersize-zz.bytes))/(float)ao_data.bps; |
954 | 498 audio_delay_method=0; // fallback if not supported |
499 } | |
3095 | 500 return ((float)ao_data.buffersize)/(float)ao_data.bps; |
954 | 501 } |