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
26 #ifdef HAVE_SOUNDCARD_H
27 #include <soundcard.h>
29 #include <sys/soundcard.h>
33 #include <sys/ioctl.h>
34 #ifdef HAVE_SYS_MMAN_H
39 #define AUDIO_BLOCK_SIZE 4096
45 int frame_size; /* in bytes ! */
48 uint8_t buffer[AUDIO_BLOCK_SIZE];
52 static int audio_open(AudioData *s, int is_output, const char *audio_device)
56 char *flip = getenv("AUDIO_FLIP_LEFT");
59 audio_fd = open(audio_device, O_WRONLY);
61 audio_fd = open(audio_device, O_RDONLY);
63 av_log(NULL, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
67 if (flip && *flip == '1') {
71 /* non blocking mode */
73 fcntl(audio_fd, F_SETFL, O_NONBLOCK);
75 s->frame_size = AUDIO_BLOCK_SIZE;
77 tmp = (NB_FRAGMENTS << 16) | FRAGMENT_BITS;
78 err = ioctl(audio_fd, SNDCTL_DSP_SETFRAGMENT, &tmp);
80 perror("SNDCTL_DSP_SETFRAGMENT");
84 /* select format : favour native format */
85 err = ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
87 #ifdef WORDS_BIGENDIAN
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(NULL, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
117 err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
119 av_log(NULL, 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(NULL, AV_LOG_ERROR, "SNDCTL_DSP_STEREO: %s\n", strerror(errno));
132 tmp = s->sample_rate;
133 err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
135 av_log(NULL, AV_LOG_ERROR, "SNDCTL_DSP_SPEED: %s\n", strerror(errno));
138 s->sample_rate = tmp; /* store real sample rate */
147 static int audio_close(AudioData *s)
153 /* sound output support */
154 static int audio_write_header(AVFormatContext *s1)
156 AudioData *s = s1->priv_data;
161 s->sample_rate = st->codec->sample_rate;
162 s->channels = st->codec->channels;
163 ret = audio_open(s, 1, s1->filename);
171 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
173 AudioData *s = s1->priv_data;
176 uint8_t *buf= pkt->data;
179 len = AUDIO_BLOCK_SIZE - s->buffer_ptr;
182 memcpy(s->buffer + s->buffer_ptr, buf, len);
183 s->buffer_ptr += len;
184 if (s->buffer_ptr >= AUDIO_BLOCK_SIZE) {
186 ret = write(s->fd, s->buffer, AUDIO_BLOCK_SIZE);
189 if (ret < 0 && (errno != EAGAIN && errno != EINTR))
200 static int audio_write_trailer(AVFormatContext *s1)
202 AudioData *s = s1->priv_data;
210 static int audio_read_header(AVFormatContext *s1, AVFormatParameters *ap)
212 AudioData *s = s1->priv_data;
216 if (ap->sample_rate <= 0 || ap->channels <= 0)
219 st = av_new_stream(s1, 0);
221 return AVERROR(ENOMEM);
223 s->sample_rate = ap->sample_rate;
224 s->channels = ap->channels;
226 ret = audio_open(s, 0, s1->filename);
232 /* take real parameters */
233 st->codec->codec_type = CODEC_TYPE_AUDIO;
234 st->codec->codec_id = s->codec_id;
235 st->codec->sample_rate = s->sample_rate;
236 st->codec->channels = s->channels;
238 av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
242 static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
244 AudioData *s = s1->priv_data;
247 struct audio_buf_info abufi;
249 if (av_new_packet(pkt, s->frame_size) < 0)
256 tv.tv_usec = 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
261 /* This will block until data is available or we get a timeout */
262 (void) select(s->fd + 1, &fds, 0, 0, &tv);
264 ret = read(s->fd, pkt->data, pkt->size);
267 if (ret == -1 && (errno == EAGAIN || errno == EINTR)) {
270 pkt->pts = av_gettime();
273 if (!(ret == 0 || (ret == -1 && (errno == EAGAIN || errno == EINTR)))) {
280 /* compute pts of the start of the packet */
281 cur_time = av_gettime();
283 if (ioctl(s->fd, SNDCTL_DSP_GETISPACE, &abufi) == 0) {
284 bdelay += abufi.bytes;
286 /* subtract time represented by the number of bytes in the audio fifo */
287 cur_time -= (bdelay * 1000000LL) / (s->sample_rate * s->channels);
289 /* convert to wanted units */
292 if (s->flip_left && s->channels == 2) {
294 short *p = (short *) pkt->data;
296 for (i = 0; i < ret; i += 4) {
304 static int audio_read_close(AVFormatContext *s1)
306 AudioData *s = s1->priv_data;
312 #ifdef CONFIG_OSS_DEMUXER
313 AVInputFormat oss_demuxer = {
315 "audio grab and output",
321 .flags = AVFMT_NOFILE,
325 #ifdef CONFIG_OSS_MUXER
326 AVOutputFormat oss_muxer = {
328 "audio grab and output",
332 /* XXX: we make the assumption that the soundcard accepts this format */
333 /* XXX: find better solution with "preinit" method, needed also in
335 #ifdef WORDS_BIGENDIAN
344 .flags = AVFMT_NOFILE,