1/*
2 * Linux audio play and grab interface
3 * Copyright (c) 2000, 2001 Fabrice Bellard
4 *
5 * This file is part of Libav.
6 *
7 * Libav 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 * Libav 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 Libav; 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 "libavutil/opt.h"
41#include "libavcodec/avcodec.h"
42#include "libavformat/avformat.h"
43#include "libavformat/internal.h"
44
45#define AUDIO_BLOCK_SIZE 4096
46
47typedef struct {
48    AVClass *class;
49    int fd;
50    int sample_rate;
51    int channels;
52    int frame_size; /* in bytes ! */
53    enum CodecID codec_id;
54    unsigned int flip_left : 1;
55    uint8_t buffer[AUDIO_BLOCK_SIZE];
56    int buffer_ptr;
57} AudioData;
58
59static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
60{
61    AudioData *s = s1->priv_data;
62    int audio_fd;
63    int tmp, err;
64    char *flip = getenv("AUDIO_FLIP_LEFT");
65
66    if (is_output)
67        audio_fd = open(audio_device, O_WRONLY);
68    else
69        audio_fd = open(audio_device, O_RDONLY);
70    if (audio_fd < 0) {
71        av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
72        return AVERROR(EIO);
73    }
74
75    if (flip && *flip == '1') {
76        s->flip_left = 1;
77    }
78
79    /* non blocking mode */
80    if (!is_output)
81        fcntl(audio_fd, F_SETFL, O_NONBLOCK);
82
83    s->frame_size = AUDIO_BLOCK_SIZE;
84
85    /* select format : favour native format */
86    err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
87
88#if HAVE_BIGENDIAN
89    if (tmp & AFMT_S16_BE) {
90        tmp = AFMT_S16_BE;
91    } else if (tmp & AFMT_S16_LE) {
92        tmp = AFMT_S16_LE;
93    } else {
94        tmp = 0;
95    }
96#else
97    if (tmp & AFMT_S16_LE) {
98        tmp = AFMT_S16_LE;
99    } else if (tmp & AFMT_S16_BE) {
100        tmp = AFMT_S16_BE;
101    } else {
102        tmp = 0;
103    }
104#endif
105
106    switch(tmp) {
107    case AFMT_S16_LE:
108        s->codec_id = CODEC_ID_PCM_S16LE;
109        break;
110    case AFMT_S16_BE:
111        s->codec_id = CODEC_ID_PCM_S16BE;
112        break;
113    default:
114        av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
115        close(audio_fd);
116        return AVERROR(EIO);
117    }
118    err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
119    if (err < 0) {
120        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
121        goto fail;
122    }
123
124    tmp = (s->channels == 2);
125    err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
126    if (err < 0) {
127        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
128        goto fail;
129    }
130
131    tmp = s->sample_rate;
132    err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
133    if (err < 0) {
134        av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
135        goto fail;
136    }
137    s->sample_rate = tmp; /* store real sample rate */
138    s->fd = audio_fd;
139
140    return 0;
141 fail:
142    close(audio_fd);
143    return AVERROR(EIO);
144}
145
146static int audio_close(AudioData *s)
147{
148    close(s->fd);
149    return 0;
150}
151
152/* sound output support */
153static int audio_write_header(AVFormatContext *s1)
154{
155    AudioData *s = s1->priv_data;
156    AVStream *st;
157    int ret;
158
159    st = s1->streams[0];
160    s->sample_rate = st->codec->sample_rate;
161    s->channels = st->codec->channels;
162    ret = audio_open(s1, 1, s1->filename);
163    if (ret < 0) {
164        return AVERROR(EIO);
165    } else {
166        return 0;
167    }
168}
169
170static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
171{
172    AudioData *s = s1->priv_data;
173    int len, ret;
174    int size= pkt->size;
175    uint8_t *buf= pkt->data;
176
177    while (size > 0) {
178        len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
179        memcpy(s->buffer + s->buffer_ptr, buf, len);
180        s->buffer_ptr += len;
181        if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
182            for(;;) {
183                ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
184                if (ret > 0)
185                    break;
186                if (ret < 0 && (errno != EAGAIN && errno != EINTR))
187                    return AVERROR(EIO);
188            }
189            s->buffer_ptr = 0;
190        }
191        buf += len;
192        size -= len;
193    }
194    return 0;
195}
196
197static int audio_write_trailer(AVFormatContext *s1)
198{
199    AudioData *s = s1->priv_data;
200
201    audio_close(s);
202    return 0;
203}
204
205/* grab support */
206
207static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
208{
209    AudioData *s = s1->priv_data;
210    AVStream *st;
211    int ret;
212
213    st = avformat_new_stream(s1, NULL);
214    if (!st) {
215        return AVERROR(ENOMEM);
216    }
217
218    ret = audio_open(s1, 0, s1->filename);
219    if (ret < 0) {
220        return AVERROR(EIO);
221    }
222
223    /* take real parameters */
224    st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
225    st->codec->codec_id = s->codec_id;
226    st->codec->sample_rate = s->sample_rate;
227    st->codec->channels = s->channels;
228
229    avpriv_set_pts_info(st, 64, 1, 1000000);  /* 64 bits pts in us */
230    return 0;
231}
232
233static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
234{
235    AudioData *s = s1->priv_data;
236    int ret, bdelay;
237    int64_t cur_time;
238    struct audio_buf_info abufi;
239
240    if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
241        return ret;
242
243    ret = read(s->fd, pkt->data, pkt->size);
244    if (ret <= 0){
245        av_free_packet(pkt);
246        pkt->size = 0;
247        if (ret<0)  return AVERROR(errno);
248        else        return AVERROR_EOF;
249    }
250    pkt->size = ret;
251
252    /* compute pts of the start of the packet */
253    cur_time = av_gettime();
254    bdelay = ret;
255    if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
256        bdelay += abufi.bytes;
257    }
258    /* subtract time represented by the number of bytes in the audio fifo */
259    cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
260
261    /* convert to wanted units */
262    pkt->pts = cur_time;
263
264    if (s->flip_left && s->channels == 2) {
265        int i;
266        short *p = (short *) pkt->data;
267
268        for (i = 0; i < ret; i += 4) {
269            *p = ~*p;
270            p += 2;
271        }
272    }
273    return 0;
274}
275
276static int audio_read_close(AVFormatContext *s1)
277{
278    AudioData *s = s1->priv_data;
279
280    audio_close(s);
281    return 0;
282}
283
284#if CONFIG_OSS_INDEV
285static const AVOption options[] = {
286    { "sample_rate", "", offsetof(AudioData, sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
287    { "channels",    "", offsetof(AudioData, channels),    AV_OPT_TYPE_INT, {.dbl = 2},     1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
288    { NULL },
289};
290
291static const AVClass oss_demuxer_class = {
292    .class_name     = "OSS demuxer",
293    .item_name      = av_default_item_name,
294    .option         = options,
295    .version        = LIBAVUTIL_VERSION_INT,
296};
297
298AVInputFormat ff_oss_demuxer = {
299    .name           = "oss",
300    .long_name      = NULL_IF_CONFIG_SMALL("Open Sound System capture"),
301    .priv_data_size = sizeof(AudioData),
302    .read_header    = audio_read_header,
303    .read_packet    = audio_read_packet,
304    .read_close     = audio_read_close,
305    .flags          = AVFMT_NOFILE,
306    .priv_class     = &oss_demuxer_class,
307};
308#endif
309
310#if CONFIG_OSS_OUTDEV
311AVOutputFormat ff_oss_muxer = {
312    .name           = "oss",
313    .long_name      = NULL_IF_CONFIG_SMALL("Open Sound System playback"),
314    .priv_data_size = 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    .audio_codec    = AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
319    .video_codec    = CODEC_ID_NONE,
320    .write_header   = audio_write_header,
321    .write_packet   = audio_write_packet,
322    .write_trailer  = audio_write_trailer,
323    .flags          = AVFMT_NOFILE,
324};
325#endif
326