avdevice: Add missing header for NULL_IF_CONFIG_SMALL
[libav.git] / libavdevice / alsa_enc.c
1 /*
2 * ALSA input and output
3 * Copyright (c) 2007 Luca Abeni ( lucabe72 email it )
4 * Copyright (c) 2007 Benoit Fouet ( benoit fouet free fr )
5 *
6 * This file is part of Libav.
7 *
8 * Libav is free software; you can redistribute it and/or
9 * modify it under the terms of the GNU Lesser General Public
10 * License as published by the Free Software Foundation; either
11 * version 2.1 of the License, or (at your option) any later version.
12 *
13 * Libav is distributed in the hope that it will be useful,
14 * but WITHOUT ANY WARRANTY; without even the implied warranty of
15 * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
16 * Lesser General Public License for more details.
17 *
18 * You should have received a copy of the GNU Lesser General Public
19 * License along with Libav; if not, write to the Free Software
20 * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
21 */
22
23 /**
24 * @file
25 * ALSA input and output: output
26 * @author Luca Abeni ( lucabe72 email it )
27 * @author Benoit Fouet ( benoit fouet free fr )
28 *
29 * This avdevice encoder allows to play audio to an ALSA (Advanced Linux
30 * Sound Architecture) device.
31 *
32 * The filename parameter is the name of an ALSA PCM device capable of
33 * capture, for example "default" or "plughw:1"; see the ALSA documentation
34 * for naming conventions. The empty string is equivalent to "default".
35 *
36 * The playback period is set to the lower value available for the device,
37 * which gives a low latency suitable for real-time playback.
38 */
39
40 #include <alsa/asoundlib.h>
41
42 #include "libavutil/internal.h"
43
44 #include "libavformat/avformat.h"
45
46 #include "alsa.h"
47
48 static av_cold int audio_write_header(AVFormatContext *s1)
49 {
50 AlsaData *s = s1->priv_data;
51 AVStream *st;
52 unsigned int sample_rate;
53 enum AVCodecID codec_id;
54 int res;
55
56 st = s1->streams[0];
57 sample_rate = st->codec->sample_rate;
58 codec_id = st->codec->codec_id;
59 res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
60 st->codec->channels, &codec_id);
61 if (sample_rate != st->codec->sample_rate) {
62 av_log(s1, AV_LOG_ERROR,
63 "sample rate %d not available, nearest is %d\n",
64 st->codec->sample_rate, sample_rate);
65 goto fail;
66 }
67
68 return res;
69
70 fail:
71 snd_pcm_close(s->h);
72 return AVERROR(EIO);
73 }
74
75 static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
76 {
77 AlsaData *s = s1->priv_data;
78 int res;
79 int size = pkt->size;
80 uint8_t *buf = pkt->data;
81
82 size /= s->frame_size;
83 if (s->reorder_func) {
84 if (size > s->reorder_buf_size)
85 if (ff_alsa_extend_reorder_buf(s, size))
86 return AVERROR(ENOMEM);
87 s->reorder_func(buf, s->reorder_buf, size);
88 buf = s->reorder_buf;
89 }
90 while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
91 if (res == -EAGAIN) {
92
93 return AVERROR(EAGAIN);
94 }
95
96 if (ff_alsa_xrun_recover(s1, res) < 0) {
97 av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
98 snd_strerror(res));
99
100 return AVERROR(EIO);
101 }
102 }
103
104 return 0;
105 }
106
107 AVOutputFormat ff_alsa_muxer = {
108 .name = "alsa",
109 .long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
110 .priv_data_size = sizeof(AlsaData),
111 .audio_codec = DEFAULT_CODEC_ID,
112 .video_codec = AV_CODEC_ID_NONE,
113 .write_header = audio_write_header,
114 .write_packet = audio_write_packet,
115 .write_trailer = ff_alsa_close,
116 .flags = AVFMT_NOFILE,
117 };