Add missing stdint.h #include to headers that use it.
[libav.git] / libavdevice / audio.c
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1/*
2 * Linux audio play and grab interface
19720f15 3 * Copyright (c) 2000, 2001 Fabrice Bellard.
de6d9b64 4 *
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5 * This file is part of FFmpeg.
6 *
7 * FFmpeg is free software; you can redistribute it and/or
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8 * modify it under the terms of the GNU Lesser General Public
9 * License as published by the Free Software Foundation; either
b78e7197 10 * version 2.1 of the License, or (at your option) any later version.
de6d9b64 11 *
b78e7197 12 * FFmpeg is distributed in the hope that it will be useful,
de6d9b64 13 * but WITHOUT ANY WARRANTY; without even the implied warranty of
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14 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15 * Lesser General Public License for more details.
de6d9b64 16 *
19720f15 17 * You should have received a copy of the GNU Lesser General Public
b78e7197 18 * License along with FFmpeg; if not, write to the Free Software
5509bffa 19 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
de6d9b64 20 */
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21#include "avformat.h"
22
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23#include <stdlib.h>
24#include <stdio.h>
25#include <string.h>
74476a3a 26#ifdef HAVE_SOUNDCARD_H
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27#include <soundcard.h>
28#else
b13788c5 29#include <sys/soundcard.h>
8c802695 30#endif
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31#include <unistd.h>
32#include <fcntl.h>
33#include <sys/ioctl.h>
34#include <sys/mman.h>
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35#include <sys/time.h>
36
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37#define AUDIO_BLOCK_SIZE 4096
38
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39typedef struct {
40 int fd;
4972b26f 41 int sample_rate;
de6d9b64 42 int channels;
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43 int frame_size; /* in bytes ! */
44 int codec_id;
1de1cce2 45 int flip_left : 1;
0c1a9eda 46 uint8_t buffer[AUDIO_BLOCK_SIZE];
4972b26f 47 int buffer_ptr;
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48} AudioData;
49
7f172339 50static int audio_open(AudioData *s, int is_output, const char *audio_device)
de6d9b64 51{
4972b26f 52 int audio_fd;
de6d9b64 53 int tmp, err;
1de1cce2 54 char *flip = getenv("AUDIO_FLIP_LEFT");
de6d9b64 55
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56 if (is_output)
57 audio_fd = open(audio_device, O_WRONLY);
de6d9b64 58 else
4972b26f 59 audio_fd = open(audio_device, O_RDONLY);
de6d9b64 60 if (audio_fd < 0) {
9f74582c 61 av_log(NULL, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
6f3e0b21 62 return AVERROR(EIO);
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63 }
64
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65 if (flip && *flip == '1') {
66 s->flip_left = 1;
67 }
68
de6d9b64 69 /* non blocking mode */
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70 if (!is_output)
71 fcntl(audio_fd, F_SETFL, O_NONBLOCK);
de6d9b64 72
4972b26f 73 s->frame_size = AUDIO_BLOCK_SIZE;
de6d9b64 74#if 0
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75 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
76 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
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77 if (err < 0) {
78 perror("SNDCTL_DSP_SETFRAGMENT");
79 }
80#endif
81
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82 /* select format : favour native format */
83 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
115329f1 84
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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:
bc874dae 111 av_log(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
4972b26f 112 close(audio_fd);
6f3e0b21 113 return AVERROR(EIO);
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114 }
115 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
de6d9b64 116 if (err < 0) {
9f74582c 117 av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
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118 goto fail;
119 }
115329f1 120
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121 tmp = (s->channels == 2);
122 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
de6d9b64 123 if (err < 0) {
9f74582c 124 av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
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125 goto fail;
126 }
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127 if (tmp)
128 s->channels = 2;
115329f1 129
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130 tmp = s->sample_rate;
131 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
de6d9b64 132 if (err < 0) {
9f74582c 133 av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
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134 goto fail;
135 }
4972b26f 136 s->sample_rate = tmp; /* store real sample rate */
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137 s->fd = audio_fd;
138
139 return 0;
140 fail:
141 close(audio_fd);
6f3e0b21 142 return AVERROR(EIO);
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143}
144
4972b26f 145static int audio_close(AudioData *s)
de6d9b64 146{
de6d9b64 147 close(s->fd);
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148 return 0;
149}
150
151/* sound output support */
152static int audio_write_header(AVFormatContext *s1)
153{
c9a65ca8 154 AudioData *s = s1->priv_data;
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155 AVStream *st;
156 int ret;
157
4972b26f 158 st = s1->streams[0];
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159 s->sample_rate = st->codec->sample_rate;
160 s->channels = st->codec->channels;
7bb7ad0e 161 ret = audio_open(s, 1, s1->filename);
4972b26f 162 if (ret < 0) {
6f3e0b21 163 return AVERROR(EIO);
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164 } else {
165 return 0;
166 }
167}
168
e928649b 169static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
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170{
171 AudioData *s = s1->priv_data;
172 int len, ret;
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173 int size= pkt->size;
174 uint8_t *buf= pkt->data;
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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);
4364a3e0 185 if (ret > 0)
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186 break;
187 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
6f3e0b21 188 return AVERROR(EIO);
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189 }
190 s->buffer_ptr = 0;
191 }
192 buf += len;
193 size -= len;
194 }
195 return 0;
196}
197
198static int audio_write_trailer(AVFormatContext *s1)
199{
200 AudioData *s = s1->priv_data;
201
202 audio_close(s);
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203 return 0;
204}
205
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206/* grab support */
207
208static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
209{
c9a65ca8 210 AudioData *s = s1->priv_data;
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211 AVStream *st;
212 int ret;
213
c04c3282 214 if (ap->sample_rate <= 0 || ap->channels <= 0)
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215 return -1;
216
c9a65ca8 217 st = av_new_stream(s1, 0);
4972b26f 218 if (!st) {
8fa36ae0 219 return AVERROR(ENOMEM);
4972b26f 220 }
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221 s->sample_rate = ap->sample_rate;
222 s->channels = ap->channels;
223
cc58300e 224 ret = audio_open(s, 0, s1->filename);
4972b26f 225 if (ret < 0) {
1ea4f593 226 av_free(st);
6f3e0b21 227 return AVERROR(EIO);
4972b26f 228 }
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229
230 /* take real parameters */
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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;
45dd5c69 235
0a7b514f 236 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
45dd5c69 237 return 0;
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238}
239
240static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
241{
242 AudioData *s = s1->priv_data;
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243 int ret, bdelay;
244 int64_t cur_time;
245 struct audio_buf_info abufi;
115329f1 246
4972b26f 247 if (av_new_packet(pkt, s->frame_size) < 0)
6f3e0b21 248 return AVERROR(EIO);
4972b26f 249 for(;;) {
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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
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262 ret = read(s->fd, pkt->data, pkt->size);
263 if (ret > 0)
264 break;
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265 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
266 av_free_packet(pkt);
267 pkt->size = 0;
0a7b514f 268 pkt->pts = av_gettime();
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269 return 0;
270 }
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271 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
272 av_free_packet(pkt);
6f3e0b21 273 return AVERROR(EIO);
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274 }
275 }
276 pkt->size = ret;
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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 }
52b541ad 284 /* subtract time represented by the number of bytes in the audio fifo */
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285 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
286
287 /* convert to wanted units */
0a7b514f 288 pkt->pts = cur_time;
45dd5c69 289
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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 }
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299 return 0;
300}
301
302static int audio_read_close(AVFormatContext *s1)
303{
304 AudioData *s = s1->priv_data;
305
306 audio_close(s);
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307 return 0;
308}
309
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310#ifdef CONFIG_OSS_DEMUXER
311AVInputFormat oss_demuxer = {
1156c6b0 312 "oss",
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313 "audio grab and output",
314 sizeof(AudioData),
315 NULL,
316 audio_read_header,
317 audio_read_packet,
318 audio_read_close,
bb76a117 319 .flags = AVFMT_NOFILE,
c9a65ca8 320};
ff70e601 321#endif
c9a65ca8 322
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323#ifdef CONFIG_OSS_MUXER
324AVOutputFormat oss_muxer = {
1156c6b0 325 "oss",
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326 "audio grab and output",
327 "",
328 "",
c9a65ca8 329 sizeof(AudioData),
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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,
bb76a117 342 .flags = AVFMT_NOFILE,
de6d9b64 343};
ff70e601 344#endif