aboutsummaryrefslogtreecommitdiffstats
path: root/libavdevice/oss_audio.c
blob: 7c4a65dda24174b8299e5549e2013db7d57e27e9 (plain) (blame)
1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
191
192
193
194
195
196
197
198
199
200
201
202
203
204
205
206
207
208
209
210
211
212
213
214
215
216
217
218
219
220
221
222
223
224
225
226
227
228
229
230
231
232
233
234
235
236
237
238
239
240
241
242
243
244
245
246
247
248
249
250
251
252
253
254
255
256
257
258
259
260
261
262
263
264
265
266
267
268
269
270
271
272
273
274
275
276
277
278
279
280
281
282
283
284
285
286
287
288
289
290
291
292
293
294
295
296
297
298
299
300
301
302
303
304
305
306
307
308
309
310
311
312
313
314
315
316
317
318
319
320
321
322
323
324
325
/*
 * Linux audio play and grab interface
 * Copyright (c) 2000, 2001 Fabrice Bellard
 *
 * This file is part of Libav.
 *
 * Libav is free software; you can redistribute it and/or
 * modify it under the terms of the GNU Lesser General Public
 * License as published by the Free Software Foundation; either
 * version 2.1 of the License, or (at your option) any later version.
 *
 * Libav is distributed in the hope that it will be useful,
 * but WITHOUT ANY WARRANTY; without even the implied warranty of
 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE.  See the GNU
 * Lesser General Public License for more details.
 *
 * You should have received a copy of the GNU Lesser General Public
 * License along with Libav; if not, write to the Free Software
 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
 */

#include "config.h"
#include <stdlib.h>
#include <stdio.h>
#include <stdint.h>
#include <string.h>
#include <errno.h>
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include <sys/time.h>
#include <sys/select.h>

#include "libavutil/log.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"

#define AUDIO_BLOCK_SIZE 4096

typedef struct {
    int fd;
    int sample_rate;
    int channels;
    int frame_size; /* in bytes ! */
    enum CodecID codec_id;
    unsigned int flip_left : 1;
    uint8_t buffer[AUDIO_BLOCK_SIZE];
    int buffer_ptr;
} AudioData;

static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
{
    AudioData *s = s1->priv_data;
    int audio_fd;
    int tmp, err;
    char *flip = getenv("AUDIO_FLIP_LEFT");

    if (is_output)
        audio_fd = open(audio_device, O_WRONLY);
    else
        audio_fd = open(audio_device, O_RDONLY);
    if (audio_fd < 0) {
        av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
        return AVERROR(EIO);
    }

    if (flip && *flip == '1') {
        s->flip_left = 1;
    }

    /* non blocking mode */
    if (!is_output)
        fcntl(audio_fd, F_SETFL, O_NONBLOCK);

    s->frame_size = AUDIO_BLOCK_SIZE;
#if 0
    tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
    err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
    if (err < 0) {
        perror("SNDCTL_DSP_SETFRAGMENT");
    }
#endif

    /* select format : favour native format */
    err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);

#if HAVE_BIGENDIAN
    if (tmp & AFMT_S16_BE) {
        tmp = AFMT_S16_BE;
    } else if (tmp & AFMT_S16_LE) {
        tmp = AFMT_S16_LE;
    } else {
        tmp = 0;
    }
#else
    if (tmp & AFMT_S16_LE) {
        tmp = AFMT_S16_LE;
    } else if (tmp & AFMT_S16_BE) {
        tmp = AFMT_S16_BE;
    } else {
        tmp = 0;
    }
#endif

    switch(tmp) {
    case AFMT_S16_LE:
        s->codec_id = CODEC_ID_PCM_S16LE;
        break;
    case AFMT_S16_BE:
        s->codec_id = CODEC_ID_PCM_S16BE;
        break;
    default:
        av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
        close(audio_fd);
        return AVERROR(EIO);
    }
    err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
    if (err < 0) {
        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
        goto fail;
    }

    tmp = (s->channels == 2);
    err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
    if (err < 0) {
        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
        goto fail;
    }

    tmp = s->sample_rate;
    err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
    if (err < 0) {
        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
        goto fail;
    }
    s->sample_rate = tmp; /* store real sample rate */
    s->fd = audio_fd;

    return 0;
 fail:
    close(audio_fd);
    return AVERROR(EIO);
}

static int audio_close(AudioData *s)
{
    close(s->fd);
    return 0;
}

/* sound output support */
static int audio_write_header(AVFormatContext *s1)
{
    AudioData *s = s1->priv_data;
    AVStream *st;
    int ret;

    st = s1->streams[0];
    s->sample_rate = st->codec->sample_rate;
    s->channels = st->codec->channels;
    ret = audio_open(s1, 1, s1->filename);
    if (ret < 0) {
        return AVERROR(EIO);
    } else {
        return 0;
    }
}

static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
{
    AudioData *s = s1->priv_data;
    int len, ret;
    int size= pkt->size;
    uint8_t *buf= pkt->data;

    while (size > 0) {
        len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
        if (len > size)
            len = size;
        memcpy(s->buffer + s->buffer_ptr, buf, len);
        s->buffer_ptr += len;
        if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
            for(;;) {
                ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
                if (ret > 0)
                    break;
                if (ret < 0 && (errno != EAGAIN && errno != EINTR))
                    return AVERROR(EIO);
            }
            s->buffer_ptr = 0;
        }
        buf += len;
        size -= len;
    }
    return 0;
}

static int audio_write_trailer(AVFormatContext *s1)
{
    AudioData *s = s1->priv_data;

    audio_close(s);
    return 0;
}

/* grab support */

static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
{
    AudioData *s = s1->priv_data;
    AVStream *st;
    int ret;

    if (ap->sample_rate <= 0 || ap->channels <= 0)
        return -1;

    st = av_new_stream(s1, 0);
    if (!st) {
        return AVERROR(ENOMEM);
    }
    s->sample_rate = ap->sample_rate;
    s->channels = ap->channels;

    ret = audio_open(s1, 0, s1->filename);
    if (ret < 0) {
        return AVERROR(EIO);
    }

    /* take real parameters */
    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
    st->codec->codec_id = s->codec_id;
    st->codec->sample_rate = s->sample_rate;
    st->codec->channels = s->channels;

    av_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
    return 0;
}

static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
{
    AudioData *s = s1->priv_data;
    int ret, bdelay;
    int64_t cur_time;
    struct audio_buf_info abufi;

    if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
        return ret;

    ret = read(s->fd, pkt->data, pkt->size);
    if (ret <= 0){
        av_free_packet(pkt);
        pkt->size = 0;
        if (ret<0)  return AVERROR(errno);
        else        return AVERROR_EOF;
    }
    pkt->size = ret;

    /* compute pts of the start of the packet */
    cur_time = av_gettime();
    bdelay = ret;
    if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
        bdelay += abufi.bytes;
    }
    /* subtract time represented by the number of bytes in the audio fifo */
    cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);

    /* convert to wanted units */
    pkt->pts = cur_time;

    if (s->flip_left && s->channels == 2) {
        int i;
        short *p = (short *) pkt->data;

        for (i = 0; i < ret; i += 4) {
            *p = ~*p;
            p += 2;
        }
    }
    return 0;
}

static int audio_read_close(AVFormatContext *s1)
{
    AudioData *s = s1->priv_data;

    audio_close(s);
    return 0;
}

#if CONFIG_OSS_INDEV
AVInputFormat ff_oss_demuxer = {
    "oss",
    NULL_IF_CONFIG_SMALL("Open Sound System capture"),
    sizeof(AudioData),
    NULL,
    audio_read_header,
    audio_read_packet,
    audio_read_close,
    .flags = AVFMT_NOFILE,
};
#endif

#if CONFIG_OSS_OUTDEV
AVOutputFormat ff_oss_muxer = {
    "oss",
    NULL_IF_CONFIG_SMALL("Open Sound System playback"),
    "",
    "",
    sizeof(AudioData),
    /* XXX: we make the assumption that the soundcard accepts this format */
    /* XXX: find better solution with "preinit" method, needed also in
       other formats */
    AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
    CODEC_ID_NONE,
    audio_write_header,
    audio_write_packet,
    audio_write_trailer,
    .flags = AVFMT_NOFILE,
};
#endif