Mercurial > libavformat.hg
annotate audio.c @ 1282:faf64f0d3445 libavformat
AVISynth support, patch by Steve Lhomme % slhomme A divxcorp P com %
Original Thread:
Date: Aug 16, 2006 11:45 PM
Subject: [Ffmpeg-devel] [PATCH] AVISynth support
author | gpoirier |
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date | Tue, 29 Aug 2006 07:34:58 +0000 |
parents | d18cc9a1fd02 |
children | 0899bfe4105c |
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 | |
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17 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA |
0 | 18 */ |
19 #include "avformat.h" | |
20 | |
21 #include <stdlib.h> | |
22 #include <stdio.h> | |
23 #include <string.h> | |
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24 #ifdef __OpenBSD__ |
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25 #include <soundcard.h> |
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26 #else |
0 | 27 #include <sys/soundcard.h> |
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28 #endif |
0 | 29 #include <unistd.h> |
30 #include <fcntl.h> | |
31 #include <sys/ioctl.h> | |
32 #include <sys/mman.h> | |
33 #include <sys/time.h> | |
34 | |
35 #define AUDIO_BLOCK_SIZE 4096 | |
36 | |
37 typedef struct { | |
38 int fd; | |
39 int sample_rate; | |
40 int channels; | |
41 int frame_size; /* in bytes ! */ | |
42 int codec_id; | |
43 int flip_left : 1; | |
65 | 44 uint8_t buffer[AUDIO_BLOCK_SIZE]; |
0 | 45 int buffer_ptr; |
46 } AudioData; | |
47 | |
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48 static int audio_open(AudioData *s, int is_output, const char *audio_device) |
0 | 49 { |
50 int audio_fd; | |
51 int tmp, err; | |
52 char *flip = getenv("AUDIO_FLIP_LEFT"); | |
53 | |
54 /* open linux audio device */ | |
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55 if (!audio_device) |
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56 #ifdef __OpenBSD__ |
887 | 57 audio_device = "/dev/sound"; |
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58 #else |
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59 audio_device = "/dev/dsp"; |
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60 #endif |
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61 |
0 | 62 if (is_output) |
63 audio_fd = open(audio_device, O_WRONLY); | |
64 else | |
65 audio_fd = open(audio_device, O_RDONLY); | |
66 if (audio_fd < 0) { | |
67 perror(audio_device); | |
482 | 68 return AVERROR_IO; |
0 | 69 } |
70 | |
71 if (flip && *flip == '1') { | |
72 s->flip_left = 1; | |
73 } | |
74 | |
75 /* non blocking mode */ | |
76 if (!is_output) | |
77 fcntl(audio_fd, F_SETFL, O_NONBLOCK); | |
78 | |
79 s->frame_size = AUDIO_BLOCK_SIZE; | |
80 #if 0 | |
81 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS; | |
82 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp); | |
83 if (err < 0) { | |
84 perror("SNDCTL_DSP_SETFRAGMENT"); | |
85 } | |
86 #endif | |
87 | |
88 /* select format : favour native format */ | |
89 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp); | |
885 | 90 |
0 | 91 #ifdef WORDS_BIGENDIAN |
92 if (tmp & AFMT_S16_BE) { | |
93 tmp = AFMT_S16_BE; | |
94 } else if (tmp & AFMT_S16_LE) { | |
95 tmp = AFMT_S16_LE; | |
96 } else { | |
97 tmp = 0; | |
98 } | |
99 #else | |
100 if (tmp & AFMT_S16_LE) { | |
101 tmp = AFMT_S16_LE; | |
102 } else if (tmp & AFMT_S16_BE) { | |
103 tmp = AFMT_S16_BE; | |
104 } else { | |
105 tmp = 0; | |
106 } | |
107 #endif | |
108 | |
109 switch(tmp) { | |
110 case AFMT_S16_LE: | |
111 s->codec_id = CODEC_ID_PCM_S16LE; | |
112 break; | |
113 case AFMT_S16_BE: | |
114 s->codec_id = CODEC_ID_PCM_S16BE; | |
115 break; | |
116 default: | |
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117 av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n"); |
0 | 118 close(audio_fd); |
482 | 119 return AVERROR_IO; |
0 | 120 } |
121 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp); | |
122 if (err < 0) { | |
123 perror("SNDCTL_DSP_SETFMT"); | |
124 goto fail; | |
125 } | |
885 | 126 |
0 | 127 tmp = (s->channels == 2); |
128 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp); | |
129 if (err < 0) { | |
130 perror("SNDCTL_DSP_STEREO"); | |
131 goto fail; | |
132 } | |
133 if (tmp) | |
134 s->channels = 2; | |
885 | 135 |
0 | 136 tmp = s->sample_rate; |
137 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp); | |
138 if (err < 0) { | |
139 perror("SNDCTL_DSP_SPEED"); | |
140 goto fail; | |
141 } | |
142 s->sample_rate = tmp; /* store real sample rate */ | |
143 s->fd = audio_fd; | |
144 | |
145 return 0; | |
146 fail: | |
147 close(audio_fd); | |
482 | 148 return AVERROR_IO; |
0 | 149 } |
150 | |
151 static int audio_close(AudioData *s) | |
152 { | |
153 close(s->fd); | |
154 return 0; | |
155 } | |
156 | |
157 /* sound output support */ | |
158 static int audio_write_header(AVFormatContext *s1) | |
159 { | |
160 AudioData *s = s1->priv_data; | |
161 AVStream *st; | |
162 int ret; | |
163 | |
164 st = s1->streams[0]; | |
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165 s->sample_rate = st->codec->sample_rate; |
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166 s->channels = st->codec->channels; |
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167 ret = audio_open(s, 1, NULL); |
0 | 168 if (ret < 0) { |
482 | 169 return AVERROR_IO; |
0 | 170 } else { |
171 return 0; | |
172 } | |
173 } | |
174 | |
468 | 175 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt) |
0 | 176 { |
177 AudioData *s = s1->priv_data; | |
178 int len, ret; | |
468 | 179 int size= pkt->size; |
180 uint8_t *buf= pkt->data; | |
0 | 181 |
182 while (size > 0) { | |
183 len = AUDIO_BLOCK_SIZE - s->buffer_ptr; | |
184 if (len > size) | |
185 len = size; | |
186 memcpy(s->buffer + s->buffer_ptr, buf, len); | |
187 s->buffer_ptr += len; | |
188 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) { | |
189 for(;;) { | |
190 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE); | |
191 if (ret > 0) | |
192 break; | |
193 if (ret < 0 && (errno != EAGAIN && errno != EINTR)) | |
482 | 194 return AVERROR_IO; |
0 | 195 } |
196 s->buffer_ptr = 0; | |
197 } | |
198 buf += len; | |
199 size -= len; | |
200 } | |
201 return 0; | |
202 } | |
203 | |
204 static int audio_write_trailer(AVFormatContext *s1) | |
205 { | |
206 AudioData *s = s1->priv_data; | |
207 | |
208 audio_close(s); | |
209 return 0; | |
210 } | |
211 | |
212 /* grab support */ | |
213 | |
214 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap) | |
215 { | |
216 AudioData *s = s1->priv_data; | |
217 AVStream *st; | |
218 int ret; | |
219 | |
1003 | 220 if (ap->sample_rate <= 0 || ap->channels <= 0) |
0 | 221 return -1; |
222 | |
223 st = av_new_stream(s1, 0); | |
224 if (!st) { | |
225 return -ENOMEM; | |
226 } | |
227 s->sample_rate = ap->sample_rate; | |
228 s->channels = ap->channels; | |
229 | |
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230 ret = audio_open(s, 0, ap->device); |
0 | 231 if (ret < 0) { |
232 av_free(st); | |
482 | 233 return AVERROR_IO; |
0 | 234 } |
235 | |
236 /* take real parameters */ | |
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237 st->codec->codec_type = CODEC_TYPE_AUDIO; |
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238 st->codec->codec_id = s->codec_id; |
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239 st->codec->sample_rate = s->sample_rate; |
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240 st->codec->channels = s->channels; |
0 | 241 |
921 | 242 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */ |
0 | 243 return 0; |
244 } | |
245 | |
246 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt) | |
247 { | |
248 AudioData *s = s1->priv_data; | |
249 int ret, bdelay; | |
250 int64_t cur_time; | |
251 struct audio_buf_info abufi; | |
885 | 252 |
0 | 253 if (av_new_packet(pkt, s->frame_size) < 0) |
482 | 254 return AVERROR_IO; |
0 | 255 for(;;) { |
56
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256 struct timeval tv; |
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257 fd_set fds; |
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258 |
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259 tv.tv_sec = 0; |
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260 tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */ |
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261 |
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262 FD_ZERO(&fds); |
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263 FD_SET(s->fd, &fds); |
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264 |
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265 /* This will block until data is available or we get a timeout */ |
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266 (void) select(s->fd + 1, &fds, 0, 0, &tv); |
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267 |
0 | 268 ret = read(s->fd, pkt->data, pkt->size); |
269 if (ret > 0) | |
270 break; | |
271 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) { | |
272 av_free_packet(pkt); | |
273 pkt->size = 0; | |
921 | 274 pkt->pts = av_gettime(); |
0 | 275 return 0; |
276 } | |
277 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) { | |
278 av_free_packet(pkt); | |
482 | 279 return AVERROR_IO; |
0 | 280 } |
281 } | |
282 pkt->size = ret; | |
283 | |
284 /* compute pts of the start of the packet */ | |
285 cur_time = av_gettime(); | |
286 bdelay = ret; | |
287 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) { | |
288 bdelay += abufi.bytes; | |
289 } | |
290 /* substract time represented by the number of bytes in the audio fifo */ | |
291 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels); | |
292 | |
293 /* convert to wanted units */ | |
921 | 294 pkt->pts = cur_time; |
0 | 295 |
296 if (s->flip_left && s->channels == 2) { | |
297 int i; | |
298 short *p = (short *) pkt->data; | |
299 | |
300 for (i = 0; i < ret; i += 4) { | |
301 *p = ~*p; | |
302 p += 2; | |
303 } | |
304 } | |
305 return 0; | |
306 } | |
307 | |
308 static int audio_read_close(AVFormatContext *s1) | |
309 { | |
310 AudioData *s = s1->priv_data; | |
311 | |
312 audio_close(s); | |
313 return 0; | |
314 } | |
315 | |
1169 | 316 #ifdef CONFIG_AUDIO_DEMUXER |
317 AVInputFormat audio_demuxer = { | |
0 | 318 "audio_device", |
319 "audio grab and output", | |
320 sizeof(AudioData), | |
321 NULL, | |
322 audio_read_header, | |
323 audio_read_packet, | |
324 audio_read_close, | |
325 .flags = AVFMT_NOFILE, | |
326 }; | |
1169 | 327 #endif |
0 | 328 |
1169 | 329 #ifdef CONFIG_AUDIO_MUXER |
330 AVOutputFormat audio_muxer = { | |
0 | 331 "audio_device", |
332 "audio grab and output", | |
333 "", | |
334 "", | |
335 sizeof(AudioData), | |
336 /* XXX: we make the assumption that the soundcard accepts this format */ | |
337 /* XXX: find better solution with "preinit" method, needed also in | |
338 other formats */ | |
339 #ifdef WORDS_BIGENDIAN | |
340 CODEC_ID_PCM_S16BE, | |
341 #else | |
342 CODEC_ID_PCM_S16LE, | |
343 #endif | |
344 CODEC_ID_NONE, | |
345 audio_write_header, | |
346 audio_write_packet, | |
347 audio_write_trailer, | |
348 .flags = AVFMT_NOFILE, | |
349 }; | |
1169 | 350 #endif |