Make av_read_frame with rtsp client return EINTR on interrupt
[libav.git] / libavdevice / audio.c
1 /*
2 * Linux audio play and grab interface
3 * Copyright (c) 2000, 2001 Fabrice Bellard.
4 *
5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
10 * version 2.1 of the License, or (at your option) any later version.
11 *
12 * FFmpeg is distributed in the hope that it will be useful,
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
18 * License along with FFmpeg; if not, write to the Free Software
19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20 */
21 #include "avformat.h"
22
23 #include <stdlib.h>
24 #include <stdio.h>
25 #include <string.h>
26 #ifdef HAVE_SOUNDCARD_H
27 #include <soundcard.h>
28 #else
29 #include <sys/soundcard.h>
30 #endif
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;
46 uint8_t buffer[AUDIO_BLOCK_SIZE];
47 int buffer_ptr;
48 } AudioData;
49
50 static int audio_open(AudioData *s, int is_output, const char *audio_device)
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 av_log(NULL, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
62 return AVERROR(EIO);
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);
84
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:
111 av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
112 close(audio_fd);
113 return AVERROR(EIO);
114 }
115 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
116 if (err < 0) {
117 av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
118 goto fail;
119 }
120
121 tmp = (s->channels == 2);
122 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
123 if (err < 0) {
124 av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
125 goto fail;
126 }
127 if (tmp)
128 s->channels = 2;
129
130 tmp = s->sample_rate;
131 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
132 if (err < 0) {
133 av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
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);
142 return AVERROR(EIO);
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];
159 s->sample_rate = st->codec->sample_rate;
160 s->channels = st->codec->channels;
161 ret = audio_open(s, 1, s1->filename);
162 if (ret < 0) {
163 return AVERROR(EIO);
164 } else {
165 return 0;
166 }
167 }
168
169 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
170 {
171 AudioData *s = s1->priv_data;
172 int len, ret;
173 int size= pkt->size;
174 uint8_t *buf= pkt->data;
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))
188 return AVERROR(EIO);
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
214 if (ap->sample_rate <= 0 || ap->channels <= 0)
215 return -1;
216
217 st = av_new_stream(s1, 0);
218 if (!st) {
219 return AVERROR(ENOMEM);
220 }
221 s->sample_rate = ap->sample_rate;
222 s->channels = ap->channels;
223
224 ret = audio_open(s, 0, s1->filename);
225 if (ret < 0) {
226 av_free(st);
227 return AVERROR(EIO);
228 }
229
230 /* take real parameters */
231 st->codec->codec_type = CODEC_TYPE_AUDIO;
232 st->codec->codec_id = s->codec_id;
233 st->codec->sample_rate = s->sample_rate;
234 st->codec->channels = s->channels;
235
236 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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;
246
247 if (av_new_packet(pkt, s->frame_size) < 0)
248 return AVERROR(EIO);
249 for(;;) {
250 struct timeval tv;
251 fd_set fds;
252
253 tv.tv_sec = 0;
254 tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
255
256 FD_ZERO(&fds);
257 FD_SET(s->fd, &fds);
258
259 /* This will block until data is available or we get a timeout */
260 (void) select(s->fd + 1, &fds, 0, 0, &tv);
261
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;
268 pkt->pts = av_gettime();
269 return 0;
270 }
271 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
272 av_free_packet(pkt);
273 return AVERROR(EIO);
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 */
288 pkt->pts = cur_time;
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
310 #ifdef CONFIG_OSS_DEMUXER
311 AVInputFormat oss_demuxer = {
312 "oss",
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 };
321 #endif
322
323 #ifdef CONFIG_OSS_MUXER
324 AVOutputFormat oss_muxer = {
325 "oss",
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 };
344 #endif