This fixes error handling for BeOS, removing the need for some ifdefs.
[libav.git] / libavformat / 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 /* open linux audio device */
57 if (!audio_device)
58 #ifdef __OpenBSD__
59 audio_device = "/dev/sound";
60 #else
61 audio_device = "/dev/dsp";
62 #endif
63
64 if (is_output)
65 audio_fd = open(audio_device, O_WRONLY);
66 else
67 audio_fd = open(audio_device, O_RDONLY);
68 if (audio_fd < 0) {
69 perror(audio_device);
70 return AVERROR_IO;
71 }
72
73 if (flip && *flip == '1') {
74 s->flip_left = 1;
75 }
76
77 /* non blocking mode */
78 if (!is_output)
79 fcntl(audio_fd, F_SETFL, O_NONBLOCK);
80
81 s->frame_size = AUDIO_BLOCK_SIZE;
82 #if 0
83 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
84 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
85 if (err < 0) {
86 perror("SNDCTL_DSP_SETFRAGMENT");
87 }
88 #endif
89
90 /* select format : favour native format */
91 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
92
93 #ifdef WORDS_BIGENDIAN
94 if (tmp & AFMT_S16_BE) {
95 tmp = AFMT_S16_BE;
96 } else if (tmp & AFMT_S16_LE) {
97 tmp = AFMT_S16_LE;
98 } else {
99 tmp = 0;
100 }
101 #else
102 if (tmp & AFMT_S16_LE) {
103 tmp = AFMT_S16_LE;
104 } else if (tmp & AFMT_S16_BE) {
105 tmp = AFMT_S16_BE;
106 } else {
107 tmp = 0;
108 }
109 #endif
110
111 switch(tmp) {
112 case AFMT_S16_LE:
113 s->codec_id = CODEC_ID_PCM_S16LE;
114 break;
115 case AFMT_S16_BE:
116 s->codec_id = CODEC_ID_PCM_S16BE;
117 break;
118 default:
119 av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
120 close(audio_fd);
121 return AVERROR_IO;
122 }
123 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
124 if (err < 0) {
125 perror("SNDCTL_DSP_SETFMT");
126 goto fail;
127 }
128
129 tmp = (s->channels == 2);
130 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
131 if (err < 0) {
132 perror("SNDCTL_DSP_STEREO");
133 goto fail;
134 }
135 if (tmp)
136 s->channels = 2;
137
138 tmp = s->sample_rate;
139 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
140 if (err < 0) {
141 perror("SNDCTL_DSP_SPEED");
142 goto fail;
143 }
144 s->sample_rate = tmp; /* store real sample rate */
145 s->fd = audio_fd;
146
147 return 0;
148 fail:
149 close(audio_fd);
150 return AVERROR_IO;
151 }
152
153 static int audio_close(AudioData *s)
154 {
155 close(s->fd);
156 return 0;
157 }
158
159 /* sound output support */
160 static int audio_write_header(AVFormatContext *s1)
161 {
162 AudioData *s = s1->priv_data;
163 AVStream *st;
164 int ret;
165
166 st = s1->streams[0];
167 s->sample_rate = st->codec->sample_rate;
168 s->channels = st->codec->channels;
169 ret = audio_open(s, 1, NULL);
170 if (ret < 0) {
171 return AVERROR_IO;
172 } else {
173 return 0;
174 }
175 }
176
177 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
178 {
179 AudioData *s = s1->priv_data;
180 int len, ret;
181 int size= pkt->size;
182 uint8_t *buf= pkt->data;
183
184 while (size > 0) {
185 len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
186 if (len > size)
187 len = size;
188 memcpy(s->buffer + s->buffer_ptr, buf, len);
189 s->buffer_ptr += len;
190 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
191 for(;;) {
192 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
193 if (ret > 0)
194 break;
195 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
196 return AVERROR_IO;
197 }
198 s->buffer_ptr = 0;
199 }
200 buf += len;
201 size -= len;
202 }
203 return 0;
204 }
205
206 static int audio_write_trailer(AVFormatContext *s1)
207 {
208 AudioData *s = s1->priv_data;
209
210 audio_close(s);
211 return 0;
212 }
213
214 /* grab support */
215
216 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
217 {
218 AudioData *s = s1->priv_data;
219 AVStream *st;
220 int ret;
221
222 if (ap->sample_rate <= 0 || ap->channels <= 0)
223 return -1;
224
225 st = av_new_stream(s1, 0);
226 if (!st) {
227 return AVERROR(ENOMEM);
228 }
229 s->sample_rate = ap->sample_rate;
230 s->channels = ap->channels;
231
232 ret = audio_open(s, 0, ap->device);
233 if (ret < 0) {
234 av_free(st);
235 return AVERROR_IO;
236 }
237
238 /* take real parameters */
239 st->codec->codec_type = CODEC_TYPE_AUDIO;
240 st->codec->codec_id = s->codec_id;
241 st->codec->sample_rate = s->sample_rate;
242 st->codec->channels = s->channels;
243
244 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
245 return 0;
246 }
247
248 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
249 {
250 AudioData *s = s1->priv_data;
251 int ret, bdelay;
252 int64_t cur_time;
253 struct audio_buf_info abufi;
254
255 if (av_new_packet(pkt, s->frame_size) < 0)
256 return AVERROR_IO;
257 for(;;) {
258 struct timeval tv;
259 fd_set fds;
260
261 tv.tv_sec = 0;
262 tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
263
264 FD_ZERO(&fds);
265 FD_SET(s->fd, &fds);
266
267 /* This will block until data is available or we get a timeout */
268 (void) select(s->fd + 1, &fds, 0, 0, &tv);
269
270 ret = read(s->fd, pkt->data, pkt->size);
271 if (ret > 0)
272 break;
273 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
274 av_free_packet(pkt);
275 pkt->size = 0;
276 pkt->pts = av_gettime();
277 return 0;
278 }
279 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
280 av_free_packet(pkt);
281 return AVERROR_IO;
282 }
283 }
284 pkt->size = ret;
285
286 /* compute pts of the start of the packet */
287 cur_time = av_gettime();
288 bdelay = ret;
289 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
290 bdelay += abufi.bytes;
291 }
292 /* substract time represented by the number of bytes in the audio fifo */
293 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
294
295 /* convert to wanted units */
296 pkt->pts = cur_time;
297
298 if (s->flip_left && s->channels == 2) {
299 int i;
300 short *p = (short *) pkt->data;
301
302 for (i = 0; i < ret; i += 4) {
303 *p = ~*p;
304 p += 2;
305 }
306 }
307 return 0;
308 }
309
310 static int audio_read_close(AVFormatContext *s1)
311 {
312 AudioData *s = s1->priv_data;
313
314 audio_close(s);
315 return 0;
316 }
317
318 #ifdef CONFIG_AUDIO_DEMUXER
319 AVInputFormat audio_demuxer = {
320 "audio_device",
321 "audio grab and output",
322 sizeof(AudioData),
323 NULL,
324 audio_read_header,
325 audio_read_packet,
326 audio_read_close,
327 .flags = AVFMT_NOFILE,
328 };
329 #endif
330
331 #ifdef CONFIG_AUDIO_MUXER
332 AVOutputFormat audio_muxer = {
333 "audio_device",
334 "audio grab and output",
335 "",
336 "",
337 sizeof(AudioData),
338 /* XXX: we make the assumption that the soundcard accepts this format */
339 /* XXX: find better solution with "preinit" method, needed also in
340 other formats */
341 #ifdef WORDS_BIGENDIAN
342 CODEC_ID_PCM_S16BE,
343 #else
344 CODEC_ID_PCM_S16LE,
345 #endif
346 CODEC_ID_NONE,
347 audio_write_header,
348 audio_write_packet,
349 audio_write_trailer,
350 .flags = AVFMT_NOFILE,
351 };
352 #endif