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>
38 #include "libavutil/log.h"
39 #include "libavcodec/avcodec.h"
40 #include "libavformat/avformat.h"
42 #define AUDIO_BLOCK_SIZE 4096
48 int frame_size
; /* in bytes ! */
51 uint8_t buffer
[AUDIO_BLOCK_SIZE
];
55 static int audio_open(AudioData
*s
, int is_output
, const char *audio_device
)
59 char *flip
= getenv("AUDIO_FLIP_LEFT");
62 audio_fd
= open(audio_device
, O_WRONLY
);
64 audio_fd
= open(audio_device
, O_RDONLY
);
66 av_log(NULL
, AV_LOG_ERROR
, "%s: %s\n", audio_device
, strerror(errno
));
70 if (flip
&& *flip
== '1') {
74 /* non blocking mode */
76 fcntl(audio_fd
, F_SETFL
, O_NONBLOCK
);
78 s
->frame_size
= AUDIO_BLOCK_SIZE
;
80 tmp
= (NB_FRAGMENTS
<< 16) | FRAGMENT_BITS
;
81 err
= ioctl(audio_fd
, SNDCTL_DSP_SETFRAGMENT
, &tmp
);
83 perror("SNDCTL_DSP_SETFRAGMENT");
87 /* select format : favour native format */
88 err
= ioctl(audio_fd
, SNDCTL_DSP_GETFMTS
, &tmp
);
90 #ifdef WORDS_BIGENDIAN
91 if (tmp
& AFMT_S16_BE
) {
93 } else if (tmp
& AFMT_S16_LE
) {
99 if (tmp
& AFMT_S16_LE
) {
101 } else if (tmp
& AFMT_S16_BE
) {
110 s
->codec_id
= CODEC_ID_PCM_S16LE
;
113 s
->codec_id
= CODEC_ID_PCM_S16BE
;
116 av_log(NULL
, AV_LOG_ERROR
, "Soundcard does not support 16 bit sample format\n");
120 err
=ioctl(audio_fd
, SNDCTL_DSP_SETFMT
, &tmp
);
122 av_log(NULL
, AV_LOG_ERROR
, "SNDCTL_DSP_SETFMT: %s\n", strerror(errno
));
126 tmp
= (s
->channels
== 2);
127 err
= ioctl(audio_fd
, SNDCTL_DSP_STEREO
, &tmp
);
129 av_log(NULL
, AV_LOG_ERROR
, "SNDCTL_DSP_STEREO: %s\n", strerror(errno
));
135 tmp
= s
->sample_rate
;
136 err
= ioctl(audio_fd
, SNDCTL_DSP_SPEED
, &tmp
);
138 av_log(NULL
, AV_LOG_ERROR
, "SNDCTL_DSP_SPEED: %s\n", strerror(errno
));
141 s
->sample_rate
= tmp
; /* store real sample rate */
150 static int audio_close(AudioData
*s
)
156 /* sound output support */
157 static int audio_write_header(AVFormatContext
*s1
)
159 AudioData
*s
= s1
->priv_data
;
164 s
->sample_rate
= st
->codec
->sample_rate
;
165 s
->channels
= st
->codec
->channels
;
166 ret
= audio_open(s
, 1, s1
->filename
);
174 static int audio_write_packet(AVFormatContext
*s1
, AVPacket
*pkt
)
176 AudioData
*s
= s1
->priv_data
;
179 uint8_t *buf
= pkt
->data
;
182 len
= AUDIO_BLOCK_SIZE
- s
->buffer_ptr
;
185 memcpy(s
->buffer
+ s
->buffer_ptr
, buf
, len
);
186 s
->buffer_ptr
+= len
;
187 if (s
->buffer_ptr
>= AUDIO_BLOCK_SIZE
) {
189 ret
= write(s
->fd
, s
->buffer
, AUDIO_BLOCK_SIZE
);
192 if (ret
< 0 && (errno
!= EAGAIN
&& errno
!= EINTR
))
203 static int audio_write_trailer(AVFormatContext
*s1
)
205 AudioData
*s
= s1
->priv_data
;
213 static int audio_read_header(AVFormatContext
*s1
, AVFormatParameters
*ap
)
215 AudioData
*s
= s1
->priv_data
;
219 if (ap
->sample_rate
<= 0 || ap
->channels
<= 0)
222 st
= av_new_stream(s1
, 0);
224 return AVERROR(ENOMEM
);
226 s
->sample_rate
= ap
->sample_rate
;
227 s
->channels
= ap
->channels
;
229 ret
= audio_open(s
, 0, s1
->filename
);
235 /* take real parameters */
236 st
->codec
->codec_type
= CODEC_TYPE_AUDIO
;
237 st
->codec
->codec_id
= s
->codec_id
;
238 st
->codec
->sample_rate
= s
->sample_rate
;
239 st
->codec
->channels
= s
->channels
;
241 av_set_pts_info(st
, 64, 1, 1000000); /* 64 bits pts in us */
245 static int audio_read_packet(AVFormatContext
*s1
, AVPacket
*pkt
)
247 AudioData
*s
= s1
->priv_data
;
250 struct audio_buf_info abufi
;
252 if (av_new_packet(pkt
, s
->frame_size
) < 0)
259 tv
.tv_usec
= 30 * 1000; /* 30 msecs -- a bit shorter than 1 frame at 30fps */
264 /* This will block until data is available or we get a timeout */
265 (void) select(s
->fd
+ 1, &fds
, 0, 0, &tv
);
267 ret
= read(s
->fd
, pkt
->data
, pkt
->size
);
270 if (ret
== -1 && (errno
== EAGAIN
|| errno
== EINTR
)) {
273 pkt
->pts
= av_gettime();
276 if (!(ret
== 0 || (ret
== -1 && (errno
== EAGAIN
|| errno
== EINTR
)))) {
283 /* compute pts of the start of the packet */
284 cur_time
= av_gettime();
286 if (ioctl(s
->fd
, SNDCTL_DSP_GETISPACE
, &abufi
) == 0) {
287 bdelay
+= abufi
.bytes
;
289 /* subtract time represented by the number of bytes in the audio fifo */
290 cur_time
-= (bdelay
* 1000000LL) / (s
->sample_rate
* s
->channels
);
292 /* convert to wanted units */
295 if (s
->flip_left
&& s
->channels
== 2) {
297 short *p
= (short *) pkt
->data
;
299 for (i
= 0; i
< ret
; i
+= 4) {
307 static int audio_read_close(AVFormatContext
*s1
)
309 AudioData
*s
= s1
->priv_data
;
315 #ifdef CONFIG_OSS_DEMUXER
316 AVInputFormat oss_demuxer
= {
318 NULL_IF_CONFIG_SMALL("audio grab and output"),
324 .flags
= AVFMT_NOFILE
,
328 #ifdef CONFIG_OSS_MUXER
329 AVOutputFormat oss_muxer
= {
331 NULL_IF_CONFIG_SMALL("audio grab and output"),
335 /* XXX: we make the assumption that the soundcard accepts this format */
336 /* XXX: find better solution with "preinit" method, needed also in
338 #ifdef WORDS_BIGENDIAN
347 .flags
= AVFMT_NOFILE
,