1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
|
/*
* Linux audio play and grab interface
* Copyright (c) 2000, 2001 Fabrice Bellard
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
#include "config.h"
#include <string.h>
#if HAVE_SOUNDCARD_H
#include <soundcard.h>
#else
#include <sys/soundcard.h>
#endif
#include <unistd.h>
#include <fcntl.h>
#include <sys/ioctl.h>
#include "libavutil/log.h"
#include "libavcodec/avcodec.h"
#include "libavformat/avformat.h"
#include "oss.h"
int ff_oss_audio_open(AVFormatContext *s1, int is_output,
const char *audio_device)
{
OSSAudioData *s = s1->priv_data;
int audio_fd;
int tmp, err;
char *flip = getenv("AUDIO_FLIP_LEFT");
char errbuff[128];
if (is_output)
audio_fd = avpriv_open(audio_device, O_WRONLY);
else
audio_fd = avpriv_open(audio_device, O_RDONLY);
if (audio_fd < 0) {
av_log(s1, AV_LOG_ERROR, "%s: %s\n", audio_device, strerror(errno));
return AVERROR(EIO);
}
if (flip && *flip == '1') {
s->flip_left = 1;
}
/* non blocking mode */
if (!is_output)
fcntl(audio_fd, F_SETFL, O_NONBLOCK);
s->frame_size = OSS_AUDIO_BLOCK_SIZE;
#define CHECK_IOCTL_ERROR(event) \
if (err < 0) { \
av_strerror(AVERROR(errno), errbuff, sizeof(errbuff)); \
av_log(s1, AV_LOG_ERROR, #event ": %s\n", errbuff); \
goto fail; \
}
/* select format : favour native format
* We don't CHECK_IOCTL_ERROR here because even if failed OSS still may be
* usable. If OSS is not usable the SNDCTL_DSP_SETFMTS later is going to
* fail anyway. */
(void) ioctl(audio_fd, SNDCTL_DSP_GETFMTS, &tmp);
#if HAVE_BIGENDIAN
if (tmp & AFMT_S16_BE) {
tmp = AFMT_S16_BE;
} else if (tmp & AFMT_S16_LE) {
tmp = AFMT_S16_LE;
} else {
tmp = 0;
}
#else
if (tmp & AFMT_S16_LE) {
tmp = AFMT_S16_LE;
} else if (tmp & AFMT_S16_BE) {
tmp = AFMT_S16_BE;
} else {
tmp = 0;
}
#endif
switch(tmp) {
case AFMT_S16_LE:
s->codec_id = AV_CODEC_ID_PCM_S16LE;
break;
case AFMT_S16_BE:
s->codec_id = AV_CODEC_ID_PCM_S16BE;
break;
default:
av_log(s1, AV_LOG_ERROR, "Soundcard does not support 16 bit sample format\n");
close(audio_fd);
return AVERROR(EIO);
}
err=ioctl(audio_fd, SNDCTL_DSP_SETFMT, &tmp);
CHECK_IOCTL_ERROR(SNDCTL_DSP_SETFMTS)
tmp = (s->channels == 2);
err = ioctl(audio_fd, SNDCTL_DSP_STEREO, &tmp);
CHECK_IOCTL_ERROR(SNDCTL_DSP_STEREO)
tmp = s->sample_rate;
err = ioctl(audio_fd, SNDCTL_DSP_SPEED, &tmp);
CHECK_IOCTL_ERROR(SNDCTL_DSP_SPEED)
s->sample_rate = tmp; /* store real sample rate */
s->fd = audio_fd;
return 0;
fail:
close(audio_fd);
return AVERROR(EIO);
#undef CHECK_IOCTL_ERROR
}
int ff_oss_audio_close(OSSAudioData *s)
{
close(s->fd);
return 0;
}
|
