2 * Linux audio play and grab interface
3 * Copyright (c) 2000, 2001 Fabrice Bellard
5 * This file is part of FFmpeg.
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.
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.
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
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"
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 /* select format : favour native format */
85 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
88 if (tmp & AFMT_S16_BE) {
90 } else if (tmp & AFMT_S16_LE) {
96 if (tmp & AFMT_S16_LE) {
98 } else if (tmp & AFMT_S16_BE) {
107 s->codec_id = CODEC_ID_PCM_S16LE;
110 s->codec_id = CODEC_ID_PCM_S16BE;
113 av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
117 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
119 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno));
123 tmp = (s->channels == 2);
124 err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
126 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
130 tmp = s->sample_rate;
131 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
133 av_log(s1, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
136 s->sample_rate = tmp; /* store real sample rate */
145 static int audio_close(AudioData *s)
151 /* sound output support */
152 static int audio_write_header(AVFormatContext *s1)
154 AudioData *s = s1->priv_data;
159 s->sample_rate = st->codec->sample_rate;
160 s->channels = st->codec->channels;
161 ret = audio_open(s1, 1, s1->filename);
169 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
171 AudioData *s = s1->priv_data;
174 uint8_t *buf= pkt->data;
177 len = FFMIN(AUDIO_BLOCK_SIZE - s->buffer_ptr, size);
178 memcpy(s->buffer + s->buffer_ptr, buf, len);
179 s->buffer_ptr += len;
180 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
182 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
185 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
196 static int audio_write_trailer(AVFormatContext *s1)
198 AudioData *s = s1->priv_data;
206 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
208 AudioData *s = s1->priv_data;
212 #if FF_API_FORMAT_PARAMETERS
213 if (ap->sample_rate > 0)
214 s->sample_rate = ap->sample_rate;
215 if (ap->channels > 0)
216 s->channels = ap->channels;
219 st = av_new_stream(s1, 0);
221 return AVERROR(ENOMEM);
224 ret = audio_open(s1, 0, s1->filename);
229 /* take real parameters */
230 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
231 st->codec->codec_id = s->codec_id;
232 st->codec->sample_rate = s->sample_rate;
233 st->codec->channels = s->channels;
235 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
239 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
241 AudioData *s = s1->priv_data;
244 struct audio_buf_info abufi;
246 if ((ret=av_new_packet(pkt, s->frame_size)) < 0)
249 ret = read(s->fd, pkt->data, pkt->size);
253 if (ret<0) return AVERROR(errno);
254 else return AVERROR_EOF;
258 /* compute pts of the start of the packet */
259 cur_time = av_gettime();
261 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
262 bdelay += abufi.bytes;
264 /* subtract time represented by the number of bytes in the audio fifo */
265 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
267 /* convert to wanted units */
270 if (s->flip_left && s->channels == 2) {
272 short *p = (short *) pkt->data;
274 for (i = 0; i < ret; i += 4) {
282 static int audio_read_close(AVFormatContext *s1)
284 AudioData *s = s1->priv_data;
291 static const AVOption options[] = {
292 { "sample_rate", "", offsetof(AudioData, sample_rate), FF_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
293 { "channels", "", offsetof(AudioData, channels), FF_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
297 static const AVClass oss_demuxer_class = {
298 .class_name = "OSS demuxer",
299 .item_name = av_default_item_name,
301 .version = LIBAVUTIL_VERSION_INT,
304 AVInputFormat ff_oss_demuxer = {
306 NULL_IF_CONFIG_SMALL("Open Sound System capture"),
312 .flags = AVFMT_NOFILE,
313 .priv_class = &oss_demuxer_class,
317 #if CONFIG_OSS_OUTDEV
318 AVOutputFormat ff_oss_muxer = {
320 NULL_IF_CONFIG_SMALL("Open Sound System playback"),
324 /* XXX: we make the assumption that the soundcard accepts this format */
325 /* XXX: find better solution with "preinit" method, needed also in
327 AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE),
332 .flags = AVFMT_NOFILE,