2 * Linux audio play and grab interface
3 * Copyright (c) 2000, 2001 Fabrice Bellard
5 * This file is part of Libav.
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.
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.
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
29 #include <soundcard.h>
31 #include <sys/soundcard.h>
35 #include <sys/ioctl.h>
37 #include <sys/select.h>
39 #include "libavutil/log.h"
40 #include "libavutil/opt.h"
41 #include "libavcodec/avcodec.h"
42 #include "libavformat/avformat.h"
44 #define AUDIO_BLOCK_SIZE 4096
51 int frame_size; /* in bytes ! */
52 enum CodecID codec_id;
53 unsigned int flip_left : 1;
54 uint8_t buffer[AUDIO_BLOCK_SIZE];
58 static int audio_open(AVFormatContext *s1, int is_output, const char *audio_device)
60 AudioData *s = s1->priv_data;
63 char *flip = getenv("AUDIO_FLIP_LEFT");
66 audio_fd = open(audio_device, O_WRONLY);
68 audio_fd = open(audio_device, O_RDONLY);
70 av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
74 if (flip && *flip == '1') {
78 /* non blocking mode */
80 fcntl(audio_fd, F_SETFL, O_NONBLOCK);
82 s->frame_size = AUDIO_BLOCK_SIZE;
84 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
85 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
87 perror("SNDCTL_DSP_SETFRAGMENT");
91 /* select format : favour native format */
92 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
95 if (tmp & AFMT_S16_BE) {
97 } else if (tmp & AFMT_S16_LE) {
103 if (tmp & AFMT_S16_LE) {
105 } else if (tmp & AFMT_S16_BE) {
114 s->codec_id = CODEC_ID_PCM_S16LE;
117 s->codec_id = CODEC_ID_PCM_S16BE;
120 av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
124 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
126 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
130 tmp = (s->channels == 2);
131 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
133 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
137 tmp = s->sample_rate;
138 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
140 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
143 s->sample_rate = tmp; /* store real sample rate */
152 static int audio_close(AudioData *s)
158 /* sound output support */
159 static int audio_write_header(AVFormatContext *s1)
161 AudioData *s = s1->priv_data;
166 s->sample_rate = st->codec->sample_rate;
167 s->channels = st->codec->channels;
168 ret = audio_open(s1, 1, s1->filename);
176 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
178 AudioData *s = s1->priv_data;
181 uint8_t *buf= pkt->data;
184 len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
187 memcpy(s->buffer + s->buffer_ptr, buf, len);
188 s->buffer_ptr += len;
189 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
191 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
194 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
205 static int audio_write_trailer(AVFormatContext *s1)
207 AudioData *s = s1->priv_data;
215 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
217 AudioData *s = s1->priv_data;
221 #if FF_API_FORMAT_PARAMETERS
222 if (ap->sample_rate > 0)
223 s->sample_rate = ap->sample_rate;
224 if (ap->channels > 0)
225 s->channels = ap->channels;
228 st = av_new_stream(s1, 0);
230 return AVERROR(ENOMEM);
233 ret = audio_open(s1, 0, s1->filename);
238 /* take real parameters */
239 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
240 st->codec->codec_id = s->codec_id;
241 st->codec->sample_rate = s->sample_rate;
242 st->codec->channels = s->channels;
244 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
248 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
250 AudioData *s = s1->priv_data;
253 struct audio_buf_info abufi;
255 if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
258 ret = read(s->fd, pkt->data, pkt->size);
262 if (ret<0) return AVERROR(errno);
263 else return AVERROR_EOF;
267 /* compute pts of the start of the packet */
268 cur_time = av_gettime();
270 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
271 bdelay += abufi.bytes;
273 /* subtract time represented by the number of bytes in the audio fifo */
274 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
276 /* convert to wanted units */
279 if (s->flip_left && s->channels == 2) {
281 short *p = (short *) pkt->data;
283 for (i = 0; i < ret; i += 4) {
291 static int audio_read_close(AVFormatContext *s1)
293 AudioData *s = s1->priv_data;
300 static const AVOption options[] = {
301 { "sample_rate", "", offsetof(AudioData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
302 { "channels", "", offsetof(AudioData, channels), FF_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
306 static const AVClass oss_demuxer_class = {
307 .class_name = "OSS demuxer",
308 .item_name = av_default_item_name,
310 .version = LIBAVUTIL_VERSION_INT,
313 AVInputFormat ff_oss_demuxer = {
315 NULL_IF_CONFIG_SMALL("Open Sound System capture"),
321 .flags = AVFMT_NOFILE,
322 .priv_class = &oss_demuxer_class,
326 #if CONFIG_OSS_OUTDEV
327 AVOutputFormat ff_oss_muxer = {
329 NULL_IF_CONFIG_SMALL("Open Sound System playback"),
333 /* XXX: we make the assumption that the soundcard accepts this format */
334 /* XXX: find better solution with "preinit" method, needed also in
336 AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
341 .flags = AVFMT_NOFILE,