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
28 #ifdef HAVE_SOUNDCARD_H
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 "libavcodec/avcodec.h"
41 #include "libavformat/avformat.h"
43 #define AUDIO_BLOCK_SIZE 4096
49 int frame_size
; /* in bytes ! */
50 enum CodecID codec_id
;
51 unsigned int flip_left
: 1;
52 uint8_t buffer
[AUDIO_BLOCK_SIZE
];
56 static int audio_open(AudioData
*s
, int is_output
, const char *audio_device
)
60 char *flip
= getenv("AUDIO_FLIP_LEFT");
63 audio_fd
= open(audio_device
, O_WRONLY
);
65 audio_fd
= open(audio_device
, O_RDONLY
);
67 av_log(NULL
, AV_LOG_ERROR
, "%s: %s\n", audio_device
, strerror(errno
));
71 if (flip
&& *flip
== '1') {
75 /* non blocking mode */
77 fcntl(audio_fd
, F_SETFL
, O_NONBLOCK
);
79 s
->frame_size
= AUDIO_BLOCK_SIZE
;
81 tmp
= (NB_FRAGMENTS
<< 16) | FRAGMENT_BITS
;
82 err
= ioctl(audio_fd
, SNDCTL_DSP_SETFRAGMENT
, &tmp
);
84 perror("SNDCTL_DSP_SETFRAGMENT");
88 /* select format : favour native format */
89 err
= ioctl(audio_fd
, SNDCTL_DSP_GETFMTS
, &tmp
);
91 #ifdef WORDS_BIGENDIAN
92 if (tmp
& AFMT_S16_BE
) {
94 } else if (tmp
& AFMT_S16_LE
) {
100 if (tmp
& AFMT_S16_LE
) {
102 } else if (tmp
& AFMT_S16_BE
) {
111 s
->codec_id
= CODEC_ID_PCM_S16LE
;
114 s
->codec_id
= CODEC_ID_PCM_S16BE
;
117 av_log(NULL
, AV_LOG_ERROR
, "Soundcard does not support 16 bit sample format\n");
121 err
=ioctl(audio_fd
, SNDCTL_DSP_SETFMT
, &tmp
);
123 av_log(NULL
, AV_LOG_ERROR
, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno
));
127 tmp
= (s
->channels
== 2);
128 err
= ioctl(audio_fd
, SNDCTL_DSP_STEREO
, &tmp
);
130 av_log(NULL
, AV_LOG_ERROR
, "SNDCTL_DSP_STEREO: %s\n", strerror(errno
));
134 tmp
= s
->sample_rate
;
135 err
= ioctl(audio_fd
, SNDCTL_DSP_SPEED
, &tmp
);
137 av_log(NULL
, AV_LOG_ERROR
, "SNDCTL_DSP_SPEED: %s\n", strerror(errno
));
140 s
->sample_rate
= tmp
; /* store real sample rate */
149 static int audio_close(AudioData
*s
)
155 /* sound output support */
156 static int audio_write_header(AVFormatContext
*s1
)
158 AudioData
*s
= s1
->priv_data
;
163 s
->sample_rate
= st
->codec
->sample_rate
;
164 s
->channels
= st
->codec
->channels
;
165 ret
= audio_open(s
, 1, s1
->filename
);
173 static int audio_write_packet(AVFormatContext
*s1
, AVPacket
*pkt
)
175 AudioData
*s
= s1
->priv_data
;
178 uint8_t *buf
= pkt
->data
;
181 len
= AUDIO_BLOCK_SIZE
- s
->buffer_ptr
;
184 memcpy(s
->buffer
+ s
->buffer_ptr
, buf
, len
);
185 s
->buffer_ptr
+= len
;
186 if (s
->buffer_ptr
>= AUDIO_BLOCK_SIZE
) {
188 ret
= write(s
->fd
, s
->buffer
, AUDIO_BLOCK_SIZE
);
191 if (ret
< 0 && (errno
!= EAGAIN
&& errno
!= EINTR
))
202 static int audio_write_trailer(AVFormatContext
*s1
)
204 AudioData
*s
= s1
->priv_data
;
212 static int audio_read_header(AVFormatContext
*s1
, AVFormatParameters
*ap
)
214 AudioData
*s
= s1
->priv_data
;
218 if (ap
->sample_rate
<= 0 || ap
->channels
<= 0)
221 st
= av_new_stream(s1
, 0);
223 return AVERROR(ENOMEM
);
225 s
->sample_rate
= ap
->sample_rate
;
226 s
->channels
= ap
->channels
;
228 ret
= audio_open(s
, 0, s1
->filename
);
234 /* take real parameters */
235 st
->codec
->codec_type
= CODEC_TYPE_AUDIO
;
236 st
->codec
->codec_id
= s
->codec_id
;
237 st
->codec
->sample_rate
= s
->sample_rate
;
238 st
->codec
->channels
= s
->channels
;
240 av_set_pts_info(st
, 64, 1, 1000000); /* 64 bits pts in us */
244 static int audio_read_packet(AVFormatContext
*s1
, AVPacket
*pkt
)
246 AudioData
*s
= s1
->priv_data
;
249 struct audio_buf_info abufi
;
251 if (av_new_packet(pkt
, s
->frame_size
) < 0)
258 tv
.tv_usec
= 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
263 /* This will block until data is available or we get a timeout */
264 (void) select(s
->fd
+ 1, &fds
, 0, 0, &tv
);
266 ret
= read(s
->fd
, pkt
->data
, pkt
->size
);
269 if (ret
== -1 && (errno
== EAGAIN
|| errno
== EINTR
)) {
272 pkt
->pts
= av_gettime();
275 if (!(ret
== 0 || (ret
== -1 && (errno
== EAGAIN
|| errno
== EINTR
)))) {
282 /* compute pts of the start of the packet */
283 cur_time
= av_gettime();
285 if (ioctl(s
->fd
, SNDCTL_DSP_GETISPACE
, &abufi
) == 0) {
286 bdelay
+= abufi
.bytes
;
288 /* subtract time represented by the number of bytes in the audio fifo */
289 cur_time
-= (bdelay
* 1000000LL) / (s
->sample_rate
* s
->channels
);
291 /* convert to wanted units */
294 if (s
->flip_left
&& s
->channels
== 2) {
296 short *p
= (short *) pkt
->data
;
298 for (i
= 0; i
< ret
; i
+= 4) {
306 static int audio_read_close(AVFormatContext
*s1
)
308 AudioData
*s
= s1
->priv_data
;
314 #ifdef CONFIG_OSS_DEMUXER
315 AVInputFormat oss_demuxer
= {
317 NULL_IF_CONFIG_SMALL("Open Sound System capture"),
323 .flags
= AVFMT_NOFILE
,
327 #ifdef CONFIG_OSS_MUXER
328 AVOutputFormat oss_muxer
= {
330 NULL_IF_CONFIG_SMALL("Open Sound System playback"),
334 /* XXX: we make the assumption that the soundcard accepts this format */
335 /* XXX: find better solution with "preinit" method, needed also in
337 #ifdef WORDS_BIGENDIAN
346 .flags
= AVFMT_NOFILE
,