Mercurial > libavformat.hg
annotate audio.c @ 2300:9c8cfadff191 libavformat
correct initial timestamps which have AV_NOPTS_VALUE
author | michael |
---|---|
date | Sat, 04 Aug 2007 22:46:13 +0000 |
parents | b21c2af60bc9 |
children | bbfeec27c98c |
rev | line source |
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0 | 1 /* |
2 * Linux audio play and grab interface | |
3 * Copyright (c) 2000, 2001 Fabrice Bellard. | |
4 * | |
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5 * This file is part of FFmpeg. |
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6 * |
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7 * FFmpeg is free software; you can redistribute it and/or |
0 | 8 * modify it under the terms of the GNU Lesser General Public |
9 * License as published by the Free Software Foundation; either | |
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10 * version 2.1 of the License, or (at your option) any later version. |
0 | 11 * |
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12 * FFmpeg is distributed in the hope that it will be useful, |
0 | 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of |
14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU | |
15 * Lesser General Public License for more details. | |
16 * | |
17 * You should have received a copy of the GNU Lesser General Public | |
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18 * License along with FFmpeg; if not, write to the Free Software |
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19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 20 */ |
21 #include "avformat.h" | |
22 | |
23 #include <stdlib.h> | |
24 #include <stdio.h> | |
25 #include <string.h> | |
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#include detected soundcard.h instead of testing for OpenBSD
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26 #ifdef HAVE_SOUNDCARD_H |
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OpenBSD support patch by (Jacob Meuser // jakemsr jakemsr com)
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27 #include <soundcard.h> |
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28 #else |
0 | 29 #include <sys/soundcard.h> |
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30 #endif |
0 | 31 #include <unistd.h> |
32 #include <fcntl.h> | |
33 #include <sys/ioctl.h> | |
34 #include <sys/mman.h> | |
35 #include <sys/time.h> | |
36 | |
37 #define AUDIO_BLOCK_SIZE 4096 | |
38 | |
39 typedef struct { | |
40 int fd; | |
41 int sample_rate; | |
42 int channels; | |
43 int frame_size; /* in bytes ! */ | |
44 int codec_id; | |
45 int flip_left : 1; | |
65 | 46 uint8_t buffer[AUDIO_BLOCK_SIZE]; |
0 | 47 int buffer_ptr; |
48 } AudioData; | |
49 | |
30
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50 static int audio_open(AudioData *s, int is_output, const char *audio_device) |
0 | 51 { |
52 int audio_fd; | |
53 int tmp, err; | |
54 char *flip = getenv("AUDIO_FLIP_LEFT"); | |
55 | |
56 if (is_output) | |
57 audio_fd = open(audio_device, O_WRONLY); | |
58 else | |
59 audio_fd = open(audio_device, O_RDONLY); | |
60 if (audio_fd < 0) { | |
61 perror(audio_device); | |
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62 return AVERROR(EIO); |
0 | 63 } |
64 | |
65 if (flip && *flip == '1') { | |
66 s->flip_left = 1; | |
67 } | |
68 | |
69 /* non blocking mode */ | |
70 if (!is_output) | |
71 fcntl(audio_fd, F_SETFL, O_NONBLOCK); | |
72 | |
73 s->frame_size = AUDIO_BLOCK_SIZE; | |
74 #if 0 | |
75 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS; | |
76 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp); | |
77 if (err < 0) { | |
78 perror("SNDCTL_DSP_SETFRAGMENT"); | |
79 } | |
80 #endif | |
81 | |
82 /* select format : favour native format */ | |
83 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp); | |
885 | 84 |
0 | 85 #ifdef WORDS_BIGENDIAN |
86 if (tmp & AFMT_S16_BE) { | |
87 tmp = AFMT_S16_BE; | |
88 } else if (tmp & AFMT_S16_LE) { | |
89 tmp = AFMT_S16_LE; | |
90 } else { | |
91 tmp = 0; | |
92 } | |
93 #else | |
94 if (tmp & AFMT_S16_LE) { | |
95 tmp = AFMT_S16_LE; | |
96 } else if (tmp & AFMT_S16_BE) { | |
97 tmp = AFMT_S16_BE; | |
98 } else { | |
99 tmp = 0; | |
100 } | |
101 #endif | |
102 | |
103 switch(tmp) { | |
104 case AFMT_S16_LE: | |
105 s->codec_id = CODEC_ID_PCM_S16LE; | |
106 break; | |
107 case AFMT_S16_BE: | |
108 s->codec_id = CODEC_ID_PCM_S16BE; | |
109 break; | |
110 default: | |
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av_log() patch by (Michel Bardiaux <mbardiaux at peaktime dot be>)
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111 av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n"); |
0 | 112 close(audio_fd); |
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113 return AVERROR(EIO); |
0 | 114 } |
115 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp); | |
116 if (err < 0) { | |
117 perror("SNDCTL_DSP_SETFMT"); | |
118 goto fail; | |
119 } | |
885 | 120 |
0 | 121 tmp = (s->channels == 2); |
122 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp); | |
123 if (err < 0) { | |
124 perror("SNDCTL_DSP_STEREO"); | |
125 goto fail; | |
126 } | |
127 if (tmp) | |
128 s->channels = 2; | |
885 | 129 |
0 | 130 tmp = s->sample_rate; |
131 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp); | |
132 if (err < 0) { | |
133 perror("SNDCTL_DSP_SPEED"); | |
134 goto fail; | |
135 } | |
136 s->sample_rate = tmp; /* store real sample rate */ | |
137 s->fd = audio_fd; | |
138 | |
139 return 0; | |
140 fail: | |
141 close(audio_fd); | |
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142 return AVERROR(EIO); |
0 | 143 } |
144 | |
145 static int audio_close(AudioData *s) | |
146 { | |
147 close(s->fd); | |
148 return 0; | |
149 } | |
150 | |
151 /* sound output support */ | |
152 static int audio_write_header(AVFormatContext *s1) | |
153 { | |
154 AudioData *s = s1->priv_data; | |
155 AVStream *st; | |
156 int ret; | |
157 | |
158 st = s1->streams[0]; | |
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159 s->sample_rate = st->codec->sample_rate; |
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160 s->channels = st->codec->channels; |
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fix oss audio playback, broken since r7983, pass AVFormatContext filename to audio_open since audio_device when NULL is no more set to default value
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161 ret = audio_open(s, 1, s1->filename); |
0 | 162 if (ret < 0) { |
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163 return AVERROR(EIO); |
0 | 164 } else { |
165 return 0; | |
166 } | |
167 } | |
168 | |
468 | 169 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 170 { |
171 AudioData *s = s1->priv_data; | |
172 int len, ret; | |
468 | 173 int size= pkt->size; |
174 uint8_t *buf= pkt->data; | |
0 | 175 |
176 while (size > 0) { | |
177 len = AUDIO_BLOCK_SIZE - s->buffer_ptr; | |
178 if (len > size) | |
179 len = size; | |
180 memcpy(s->buffer + s->buffer_ptr, buf, len); | |
181 s->buffer_ptr += len; | |
182 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) { | |
183 for(;;) { | |
184 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE); | |
185 if (ret > 0) | |
186 break; | |
187 if (ret < 0 && (errno != EAGAIN && errno != EINTR)) | |
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188 return AVERROR(EIO); |
0 | 189 } |
190 s->buffer_ptr = 0; | |
191 } | |
192 buf += len; | |
193 size -= len; | |
194 } | |
195 return 0; | |
196 } | |
197 | |
198 static int audio_write_trailer(AVFormatContext *s1) | |
199 { | |
200 AudioData *s = s1->priv_data; | |
201 | |
202 audio_close(s); | |
203 return 0; | |
204 } | |
205 | |
206 /* grab support */ | |
207 | |
208 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) | |
209 { | |
210 AudioData *s = s1->priv_data; | |
211 AVStream *st; | |
212 int ret; | |
213 | |
1003 | 214 if (ap->sample_rate <= 0 || ap->channels <= 0) |
0 | 215 return -1; |
216 | |
217 st = av_new_stream(s1, 0); | |
218 if (!st) { | |
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219 return AVERROR(ENOMEM); |
0 | 220 } |
221 s->sample_rate = ap->sample_rate; | |
222 s->channels = ap->channels; | |
223 | |
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224 ret = audio_open(s, 0, s1->filename); |
0 | 225 if (ret < 0) { |
226 av_free(st); | |
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227 return AVERROR(EIO); |
0 | 228 } |
229 | |
230 /* take real parameters */ | |
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231 st->codec->codec_type = CODEC_TYPE_AUDIO; |
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232 st->codec->codec_id = s->codec_id; |
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233 st->codec->sample_rate = s->sample_rate; |
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234 st->codec->channels = s->channels; |
0 | 235 |
921 | 236 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ |
0 | 237 return 0; |
238 } | |
239 | |
240 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) | |
241 { | |
242 AudioData *s = s1->priv_data; | |
243 int ret, bdelay; | |
244 int64_t cur_time; | |
245 struct audio_buf_info abufi; | |
885 | 246 |
0 | 247 if (av_new_packet(pkt, s->frame_size) < 0) |
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248 return AVERROR(EIO); |
0 | 249 for(;;) { |
56
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250 struct timeval tv; |
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251 fd_set fds; |
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252 |
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253 tv.tv_sec = 0; |
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254 tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */ |
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255 |
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256 FD_ZERO(&fds); |
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257 FD_SET(s->fd, &fds); |
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258 |
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259 /* This will block until data is available or we get a timeout */ |
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260 (void) select(s->fd + 1, &fds, 0, 0, &tv); |
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261 |
0 | 262 ret = read(s->fd, pkt->data, pkt->size); |
263 if (ret > 0) | |
264 break; | |
265 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) { | |
266 av_free_packet(pkt); | |
267 pkt->size = 0; | |
921 | 268 pkt->pts = av_gettime(); |
0 | 269 return 0; |
270 } | |
271 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) { | |
272 av_free_packet(pkt); | |
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273 return AVERROR(EIO); |
0 | 274 } |
275 } | |
276 pkt->size = ret; | |
277 | |
278 /* compute pts of the start of the packet */ | |
279 cur_time = av_gettime(); | |
280 bdelay = ret; | |
281 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) { | |
282 bdelay += abufi.bytes; | |
283 } | |
284 /* substract time represented by the number of bytes in the audio fifo */ | |
285 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels); | |
286 | |
287 /* convert to wanted units */ | |
921 | 288 pkt->pts = cur_time; |
0 | 289 |
290 if (s->flip_left && s->channels == 2) { | |
291 int i; | |
292 short *p = (short *) pkt->data; | |
293 | |
294 for (i = 0; i < ret; i += 4) { | |
295 *p = ~*p; | |
296 p += 2; | |
297 } | |
298 } | |
299 return 0; | |
300 } | |
301 | |
302 static int audio_read_close(AVFormatContext *s1) | |
303 { | |
304 AudioData *s = s1->priv_data; | |
305 | |
306 audio_close(s); | |
307 return 0; | |
308 } | |
309 | |
1169 | 310 #ifdef CONFIG_AUDIO_DEMUXER |
311 AVInputFormat audio_demuxer = { | |
0 | 312 "audio_device", |
313 "audio grab and output", | |
314 sizeof(AudioData), | |
315 NULL, | |
316 audio_read_header, | |
317 audio_read_packet, | |
318 audio_read_close, | |
319 .flags = AVFMT_NOFILE, | |
320 }; | |
1169 | 321 #endif |
0 | 322 |
1169 | 323 #ifdef CONFIG_AUDIO_MUXER |
324 AVOutputFormat audio_muxer = { | |
0 | 325 "audio_device", |
326 "audio grab and output", | |
327 "", | |
328 "", | |
329 sizeof(AudioData), | |
330 /* XXX: we make the assumption that the soundcard accepts this format */ | |
331 /* XXX: find better solution with "preinit" method, needed also in | |
332 other formats */ | |
333 #ifdef WORDS_BIGENDIAN | |
334 CODEC_ID_PCM_S16BE, | |
335 #else | |
336 CODEC_ID_PCM_S16LE, | |
337 #endif | |
338 CODEC_ID_NONE, | |
339 audio_write_header, | |
340 audio_write_packet, | |
341 audio_write_trailer, | |
342 .flags = AVFMT_NOFILE, | |
343 }; | |
1169 | 344 #endif |