Statistics
| Branch: | Revision:

ffmpeg / libavformat / audio.c @ 71e445fc

History | View | Annotate | Download (8.25 KB)

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 __OpenBSD__
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 -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