Mercurial > libavformat.hg
annotate audio.c @ 292:155ce13f1033 libavformat
mpeg TS demux API (called from RTP layer) - raw mpeg TS reader (uses fake codec MPEG2TS)
author | bellard |
---|---|
date | Wed, 29 Oct 2003 14:16:31 +0000 |
parents | 3d92f793fd67 |
children | 845f9de2c883 |
rev | line source |
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0 | 1 /* |
2 * Linux audio play and grab interface | |
3 * Copyright (c) 2000, 2001 Fabrice Bellard. | |
4 * | |
5 * This library is free software; you can redistribute it and/or | |
6 * modify it under the terms of the GNU Lesser General Public | |
7 * License as published by the Free Software Foundation; either | |
8 * version 2 of the License, or (at your option) any later version. | |
9 * | |
10 * This library is distributed in the hope that it will be useful, | |
11 * but WITHOUT ANY WARRANTY; without even the implied warranty of | |
12 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
13 * Lesser General Public License for more details. | |
14 * | |
15 * You should have received a copy of the GNU Lesser General Public | |
16 * License along with this library; if not, write to the Free Software | |
17 * Foundation, Inc., 59 Temple Place, Suite 330, Boston, MA 02111-1307 USA | |
18 */ | |
19 #include "avformat.h" | |
20 | |
21 #include <stdlib.h> | |
22 #include <stdio.h> | |
23 #include <string.h> | |
24 #include <sys/soundcard.h> | |
25 #include <unistd.h> | |
26 #include <fcntl.h> | |
27 #include <sys/ioctl.h> | |
28 #include <sys/mman.h> | |
29 #include <sys/time.h> | |
30 | |
31 #define AUDIO_BLOCK_SIZE 4096 | |
32 | |
33 typedef struct { | |
34 int fd; | |
35 int sample_rate; | |
36 int channels; | |
37 int frame_size; /* in bytes ! */ | |
38 int codec_id; | |
39 int flip_left : 1; | |
65 | 40 uint8_t buffer[AUDIO_BLOCK_SIZE]; |
0 | 41 int buffer_ptr; |
42 } AudioData; | |
43 | |
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44 static int audio_open(AudioData *s, int is_output, const char *audio_device) |
0 | 45 { |
46 int audio_fd; | |
47 int tmp, err; | |
48 char *flip = getenv("AUDIO_FLIP_LEFT"); | |
49 | |
50 /* open linux audio device */ | |
30
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51 if (!audio_device) |
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52 audio_device = "/dev/dsp"; |
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53 |
0 | 54 if (is_output) |
55 audio_fd = open(audio_device, O_WRONLY); | |
56 else | |
57 audio_fd = open(audio_device, O_RDONLY); | |
58 if (audio_fd < 0) { | |
59 perror(audio_device); | |
60 return -EIO; | |
61 } | |
62 | |
63 if (flip && *flip == '1') { | |
64 s->flip_left = 1; | |
65 } | |
66 | |
67 /* non blocking mode */ | |
68 if (!is_output) | |
69 fcntl(audio_fd, F_SETFL, O_NONBLOCK); | |
70 | |
71 s->frame_size = AUDIO_BLOCK_SIZE; | |
72 #if 0 | |
73 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS; | |
74 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp); | |
75 if (err < 0) { | |
76 perror("SNDCTL_DSP_SETFRAGMENT"); | |
77 } | |
78 #endif | |
79 | |
80 /* select format : favour native format */ | |
81 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp); | |
82 | |
83 #ifdef WORDS_BIGENDIAN | |
84 if (tmp & AFMT_S16_BE) { | |
85 tmp = AFMT_S16_BE; | |
86 } else if (tmp & AFMT_S16_LE) { | |
87 tmp = AFMT_S16_LE; | |
88 } else { | |
89 tmp = 0; | |
90 } | |
91 #else | |
92 if (tmp & AFMT_S16_LE) { | |
93 tmp = AFMT_S16_LE; | |
94 } else if (tmp & AFMT_S16_BE) { | |
95 tmp = AFMT_S16_BE; | |
96 } else { | |
97 tmp = 0; | |
98 } | |
99 #endif | |
100 | |
101 switch(tmp) { | |
102 case AFMT_S16_LE: | |
103 s->codec_id = CODEC_ID_PCM_S16LE; | |
104 break; | |
105 case AFMT_S16_BE: | |
106 s->codec_id = CODEC_ID_PCM_S16BE; | |
107 break; | |
108 default: | |
109 fprintf(stderr, "Soundcard does not support 16 bit sample format\n"); | |
110 close(audio_fd); | |
111 return -EIO; | |
112 } | |
113 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp); | |
114 if (err < 0) { | |
115 perror("SNDCTL_DSP_SETFMT"); | |
116 goto fail; | |
117 } | |
118 | |
119 tmp = (s->channels == 2); | |
120 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp); | |
121 if (err < 0) { | |
122 perror("SNDCTL_DSP_STEREO"); | |
123 goto fail; | |
124 } | |
125 if (tmp) | |
126 s->channels = 2; | |
127 | |
128 tmp = s->sample_rate; | |
129 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp); | |
130 if (err < 0) { | |
131 perror("SNDCTL_DSP_SPEED"); | |
132 goto fail; | |
133 } | |
134 s->sample_rate = tmp; /* store real sample rate */ | |
135 s->fd = audio_fd; | |
136 | |
137 return 0; | |
138 fail: | |
139 close(audio_fd); | |
140 return -EIO; | |
141 } | |
142 | |
143 static int audio_close(AudioData *s) | |
144 { | |
145 close(s->fd); | |
146 return 0; | |
147 } | |
148 | |
149 /* sound output support */ | |
150 static int audio_write_header(AVFormatContext *s1) | |
151 { | |
152 AudioData *s = s1->priv_data; | |
153 AVStream *st; | |
154 int ret; | |
155 | |
156 st = s1->streams[0]; | |
157 s->sample_rate = st->codec.sample_rate; | |
158 s->channels = st->codec.channels; | |
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159 ret = audio_open(s, 1, NULL); |
0 | 160 if (ret < 0) { |
161 return -EIO; | |
162 } else { | |
163 return 0; | |
164 } | |
165 } | |
166 | |
167 static int audio_write_packet(AVFormatContext *s1, int stream_index, | |
241 | 168 const uint8_t *buf, int size, int64_t pts) |
0 | 169 { |
170 AudioData *s = s1->priv_data; | |
171 int len, ret; | |
172 | |
173 while (size > 0) { | |
174 len = AUDIO_BLOCK_SIZE - s->buffer_ptr; | |
175 if (len > size) | |
176 len = size; | |
177 memcpy(s->buffer + s->buffer_ptr, buf, len); | |
178 s->buffer_ptr += len; | |
179 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) { | |
180 for(;;) { | |
181 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE); | |
182 if (ret > 0) | |
183 break; | |
184 if (ret < 0 && (errno != EAGAIN && errno != EINTR)) | |
185 return -EIO; | |
186 } | |
187 s->buffer_ptr = 0; | |
188 } | |
189 buf += len; | |
190 size -= len; | |
191 } | |
192 return 0; | |
193 } | |
194 | |
195 static int audio_write_trailer(AVFormatContext *s1) | |
196 { | |
197 AudioData *s = s1->priv_data; | |
198 | |
199 audio_close(s); | |
200 return 0; | |
201 } | |
202 | |
203 /* grab support */ | |
204 | |
205 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) | |
206 { | |
207 AudioData *s = s1->priv_data; | |
208 AVStream *st; | |
209 int ret; | |
210 | |
211 if (!ap || ap->sample_rate <= 0 || ap->channels <= 0) | |
212 return -1; | |
213 | |
214 st = av_new_stream(s1, 0); | |
215 if (!st) { | |
216 return -ENOMEM; | |
217 } | |
218 s->sample_rate = ap->sample_rate; | |
219 s->channels = ap->channels; | |
220 | |
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221 ret = audio_open(s, 0, ap->device); |
0 | 222 if (ret < 0) { |
223 av_free(st); | |
224 return -EIO; | |
225 } | |
226 | |
227 /* take real parameters */ | |
228 st->codec.codec_type = CODEC_TYPE_AUDIO; | |
229 st->codec.codec_id = s->codec_id; | |
230 st->codec.sample_rate = s->sample_rate; | |
231 st->codec.channels = s->channels; | |
232 | |
233 av_set_pts_info(s1, 48, 1, 1000000); /* 48 bits pts in us */ | |
234 return 0; | |
235 } | |
236 | |
237 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) | |
238 { | |
239 AudioData *s = s1->priv_data; | |
240 int ret, bdelay; | |
241 int64_t cur_time; | |
242 struct audio_buf_info abufi; | |
243 | |
244 if (av_new_packet(pkt, s->frame_size) < 0) | |
245 return -EIO; | |
246 for(;;) { | |
56
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247 struct timeval tv; |
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248 fd_set fds; |
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249 |
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250 tv.tv_sec = 0; |
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251 tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */ |
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252 |
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253 FD_ZERO(&fds); |
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254 FD_SET(s->fd, &fds); |
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255 |
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256 /* This will block until data is available or we get a timeout */ |
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257 (void) select(s->fd + 1, &fds, 0, 0, &tv); |
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258 |
0 | 259 ret = read(s->fd, pkt->data, pkt->size); |
260 if (ret > 0) | |
261 break; | |
262 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) { | |
263 av_free_packet(pkt); | |
264 pkt->size = 0; | |
265 return 0; | |
266 } | |
267 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) { | |
268 av_free_packet(pkt); | |
269 return -EIO; | |
270 } | |
271 } | |
272 pkt->size = ret; | |
273 | |
274 /* compute pts of the start of the packet */ | |
275 cur_time = av_gettime(); | |
276 bdelay = ret; | |
277 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) { | |
278 bdelay += abufi.bytes; | |
279 } | |
280 /* substract time represented by the number of bytes in the audio fifo */ | |
281 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels); | |
282 | |
283 /* convert to wanted units */ | |
284 pkt->pts = cur_time & ((1LL << 48) - 1); | |
285 | |
286 if (s->flip_left && s->channels == 2) { | |
287 int i; | |
288 short *p = (short *) pkt->data; | |
289 | |
290 for (i = 0; i < ret; i += 4) { | |
291 *p = ~*p; | |
292 p += 2; | |
293 } | |
294 } | |
295 return 0; | |
296 } | |
297 | |
298 static int audio_read_close(AVFormatContext *s1) | |
299 { | |
300 AudioData *s = s1->priv_data; | |
301 | |
302 audio_close(s); | |
303 return 0; | |
304 } | |
305 | |
306 static AVInputFormat audio_in_format = { | |
307 "audio_device", | |
308 "audio grab and output", | |
309 sizeof(AudioData), | |
310 NULL, | |
311 audio_read_header, | |
312 audio_read_packet, | |
313 audio_read_close, | |
314 .flags = AVFMT_NOFILE, | |
315 }; | |
316 | |
317 static AVOutputFormat audio_out_format = { | |
318 "audio_device", | |
319 "audio grab and output", | |
320 "", | |
321 "", | |
322 sizeof(AudioData), | |
323 /* XXX: we make the assumption that the soundcard accepts this format */ | |
324 /* XXX: find better solution with "preinit" method, needed also in | |
325 other formats */ | |
326 #ifdef WORDS_BIGENDIAN | |
327 CODEC_ID_PCM_S16BE, | |
328 #else | |
329 CODEC_ID_PCM_S16LE, | |
330 #endif | |
331 CODEC_ID_NONE, | |
332 audio_write_header, | |
333 audio_write_packet, | |
334 audio_write_trailer, | |
335 .flags = AVFMT_NOFILE, | |
336 }; | |
337 | |
338 int audio_init(void) | |
339 { | |
340 av_register_input_format(&audio_in_format); | |
341 av_register_output_format(&audio_out_format); | |
342 return 0; | |
343 } |