Add entry for AVFilterPicRef interlaced and top_field_first fields
[libav.git] / libavdevice / oss_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
22 #include "config.h"
23 #include <stdlib.h>
24 #include <stdio.h>
25 #include <stdint.h>
26 #include <string.h>
27 #include <errno.h>
28 #if HAVE_SOUNDCARD_H
29 #include <soundcard.h>
30 #else
31 #include <sys/soundcard.h>
32 #endif
33 #include <unistd.h>
34 #include <fcntl.h>
35 #include <sys/ioctl.h>
36 #include <sys/time.h>
37 #include <sys/select.h>
38
39 #include "libavutil/log.h"
40 #include "libavcodec/avcodec.h"
41 #include "libavformat/avformat.h"
42
43 #define AUDIO_BLOCK_SIZE 4096
44
45 typedef struct {
46 int fd;
47 int sample_rate;
48 int channels;
49 int frame_size; /* in bytes ! */
50 enum CodecID codec_id;
51 unsigned int flip_left : 1;
52 uint8_t buffer[AUDIO_BLOCK_SIZE];
53 int buffer_ptr;
54 } AudioData;
55
56 static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
57 {
58 AudioData *s = s1->priv_data;
59 int audio_fd;
60 int tmp, err;
61 char *flip = getenv("AUDIO_FLIP_LEFT");
62
63 if (is_output)
64 audio_fd = open(audio_device, O_WRONLY);
65 else
66 audio_fd = open(audio_device, O_RDONLY);
67 if (audio_fd < 0) {
68 av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
69 return AVERROR(EIO);
70 }
71
72 if (flip && *flip == '1') {
73 s->flip_left = 1;
74 }
75
76 /* non blocking mode */
77 if (!is_output)
78 fcntl(audio_fd, F_SETFL, O_NONBLOCK);
79
80 s->frame_size = AUDIO_BLOCK_SIZE;
81 #if 0
82 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
83 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
84 if (err < 0) {
85 perror("SNDCTL_DSP_SETFRAGMENT");
86 }
87 #endif
88
89 /* select format : favour native format */
90 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
91
92 #if HAVE_BIGENDIAN
93 if (tmp & AFMT_S16_BE) {
94 tmp = AFMT_S16_BE;
95 } else if (tmp & AFMT_S16_LE) {
96 tmp = AFMT_S16_LE;
97 } else {
98 tmp = 0;
99 }
100 #else
101 if (tmp & AFMT_S16_LE) {
102 tmp = AFMT_S16_LE;
103 } else if (tmp & AFMT_S16_BE) {
104 tmp = AFMT_S16_BE;
105 } else {
106 tmp = 0;
107 }
108 #endif
109
110 switch(tmp) {
111 case AFMT_S16_LE:
112 s->codec_id = CODEC_ID_PCM_S16LE;
113 break;
114 case AFMT_S16_BE:
115 s->codec_id = CODEC_ID_PCM_S16BE;
116 break;
117 default:
118 av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
119 close(audio_fd);
120 return AVERROR(EIO);
121 }
122 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
123 if (err < 0) {
124 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
125 goto fail;
126 }
127
128 tmp = (s->channels == 2);
129 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
130 if (err < 0) {
131 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
132 goto fail;
133 }
134
135 tmp = s->sample_rate;
136 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
137 if (err < 0) {
138 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
139 goto fail;
140 }
141 s->sample_rate = tmp; /* store real sample rate */
142 s->fd = audio_fd;
143
144 return 0;
145 fail:
146 close(audio_fd);
147 return AVERROR(EIO);
148 }
149
150 static int audio_close(AudioData *s)
151 {
152 close(s->fd);
153 return 0;
154 }
155
156 /* sound output support */
157 static int audio_write_header(AVFormatContext *s1)
158 {
159 AudioData *s = s1->priv_data;
160 AVStream *st;
161 int ret;
162
163 st = s1->streams[0];
164 s->sample_rate = st->codec->sample_rate;
165 s->channels = st->codec->channels;
166 ret = audio_open(s1, 1, s1->filename);
167 if (ret < 0) {
168 return AVERROR(EIO);
169 } else {
170 return 0;
171 }
172 }
173
174 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
175 {
176 AudioData *s = s1->priv_data;
177 int len, ret;
178 int size= pkt->size;
179 uint8_t *buf= pkt->data;
180
181 while (size > 0) {
182 len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
183 if (len > size)
184 len = size;
185 memcpy(s->buffer + s->buffer_ptr, buf, len);
186 s->buffer_ptr += len;
187 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
188 for(;;) {
189 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
190 if (ret > 0)
191 break;
192 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
193 return AVERROR(EIO);
194 }
195 s->buffer_ptr = 0;
196 }
197 buf += len;
198 size -= len;
199 }
200 return 0;
201 }
202
203 static int audio_write_trailer(AVFormatContext *s1)
204 {
205 AudioData *s = s1->priv_data;
206
207 audio_close(s);
208 return 0;
209 }
210
211 /* grab support */
212
213 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
214 {
215 AudioData *s = s1->priv_data;
216 AVStream *st;
217 int ret;
218
219 if (ap->sample_rate <= 0 || ap->channels <= 0)
220 return -1;
221
222 st = av_new_stream(s1, 0);
223 if (!st) {
224 return AVERROR(ENOMEM);
225 }
226 s->sample_rate = ap->sample_rate;
227 s->channels = ap->channels;
228
229 ret = audio_open(s1, 0, s1->filename);
230 if (ret < 0) {
231 return AVERROR(EIO);
232 }
233
234 /* take real parameters */
235 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
236 st->codec->codec_id = s->codec_id;
237 st->codec->sample_rate = s->sample_rate;
238 st->codec->channels = s->channels;
239
240 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
241 return 0;
242 }
243
244 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
245 {
246 AudioData *s = s1->priv_data;
247 int ret, bdelay;
248 int64_t cur_time;
249 struct audio_buf_info abufi;
250
251 if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
252 return ret;
253
254 ret = read(s->fd, pkt->data, pkt->size);
255 if (ret <= 0){
256 av_free_packet(pkt);
257 pkt->size = 0;
258 if (ret<0) return AVERROR(errno);
259 else return AVERROR_EOF;
260 }
261 pkt->size = ret;
262
263 /* compute pts of the start of the packet */
264 cur_time = av_gettime();
265 bdelay = ret;
266 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
267 bdelay += abufi.bytes;
268 }
269 /* subtract time represented by the number of bytes in the audio fifo */
270 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
271
272 /* convert to wanted units */
273 pkt->pts = cur_time;
274
275 if (s->flip_left && s->channels == 2) {
276 int i;
277 short *p = (short *) pkt->data;
278
279 for (i = 0; i < ret; i += 4) {
280 *p = ~*p;
281 p += 2;
282 }
283 }
284 return 0;
285 }
286
287 static int audio_read_close(AVFormatContext *s1)
288 {
289 AudioData *s = s1->priv_data;
290
291 audio_close(s);
292 return 0;
293 }
294
295 #if CONFIG_OSS_INDEV
296 AVInputFormat oss_demuxer = {
297 "oss",
298 NULL_IF_CONFIG_SMALL("Open Sound System capture"),
299 sizeof(AudioData),
300 NULL,
301 audio_read_header,
302 audio_read_packet,
303 audio_read_close,
304 .flags = AVFMT_NOFILE,
305 };
306 #endif
307
308 #if CONFIG_OSS_OUTDEV
309 AVOutputFormat oss_muxer = {
310 "oss",
311 NULL_IF_CONFIG_SMALL("Open Sound System playback"),
312 "",
313 "",
314 sizeof(AudioData),
315 /* XXX: we make the assumption that the soundcard accepts this format */
316 /* XXX: find better solution with "preinit" method, needed also in
317 other formats */
318 #if HAVE_BIGENDIAN
319 CODEC_ID_PCM_S16BE,
320 #else
321 CODEC_ID_PCM_S16LE,
322 #endif
323 CODEC_ID_NONE,
324 audio_write_header,
325 audio_write_packet,
326 audio_write_trailer,
327 .flags = AVFMT_NOFILE,
328 };
329 #endif