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 FFmpeg.
7 *
8 * FFmpeg 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 * FFmpeg 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 FFmpeg; 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 libavdevice/alsa-audio-enc.c
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 "libavformat/avformat.h"
41#include <alsa/asoundlib.h>
42
43#include "alsa-audio.h"
44
45av_cold static int audio_write_header(AVFormatContext *s1)
46{
47    AlsaData *s = s1->priv_data;
48    AVStream *st;
49    unsigned int sample_rate;
50    int codec_id;
51    int res;
52
53    st = s1->streams[0];
54    sample_rate = st->codec->sample_rate;
55    codec_id    = st->codec->codec_id;
56    res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
57        st->codec->channels, &codec_id);
58    if (sample_rate != st->codec->sample_rate) {
59        av_log(s1, AV_LOG_ERROR,
60               "sample rate %d not available, nearest is %d\n",
61               st->codec->sample_rate, sample_rate);
62        goto fail;
63    }
64
65    return res;
66
67fail:
68    snd_pcm_close(s->h);
69    return AVERROR(EIO);
70}
71
72static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
73{
74    AlsaData *s = s1->priv_data;
75    int res;
76    int size     = pkt->size;
77    uint8_t *buf = pkt->data;
78
79    while((res = snd_pcm_writei(s->h, buf, size / s->frame_size)) < 0) {
80        if (res == -EAGAIN) {
81
82            return AVERROR(EAGAIN);
83        }
84
85        if (ff_alsa_xrun_recover(s1, res) < 0) {
86            av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
87                   snd_strerror(res));
88
89            return AVERROR(EIO);
90        }
91    }
92
93    return 0;
94}
95
96AVOutputFormat alsa_muxer = {
97    "alsa",
98    NULL_IF_CONFIG_SMALL("ALSA audio output"),
99    "",
100    "",
101    sizeof(AlsaData),
102    DEFAULT_CODEC_ID,
103    CODEC_ID_NONE,
104    audio_write_header,
105    audio_write_packet,
106    ff_alsa_close,
107    .flags = AVFMT_NOFILE,
108};
109