path: root/libavdevice/alsa_dec.c (plain)
blob: c50ce715064ad20f21c7d3c41d1e3003c69fef3f

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
25	* ALSA input and output: input
26	* @author Luca Abeni ( lucabe72 email it )
27	* @author Benoit Fouet ( benoit fouet free fr )
28	* @author Nicolas George ( nicolas george normalesup org )
29	*
30	* This avdevice decoder can capture audio from an ALSA (Advanced
31	* Linux Sound Architecture) device.
32	*
33	* The filename parameter is the name of an ALSA PCM device capable of
34	* capture, for example "default" or "plughw:1"; see the ALSA documentation
35	* for naming conventions. The empty string is equivalent to "default".
36	*
37	* The capture period is set to the lower value available for the device,
38	* which gives a low latency suitable for real-time capture.
39	*
40	* The PTS are an Unix time in microsecond.
41	*
42	* Due to a bug in the ALSA library
43	* (https://bugtrack.alsa-project.org/alsa-bug/view.php?id=4308), this
44	* decoder does not work with certain ALSA plugins, especially the dsnoop
45	* plugin.
46	*/
47
48	#include <alsa/asoundlib.h>
49
50	#include "libavutil/internal.h"
51	#include "libavutil/mathematics.h"
52	#include "libavutil/opt.h"
53	#include "libavutil/time.h"
54
55	#include "libavformat/internal.h"
56
57	#include "avdevice.h"
58	#include "alsa.h"
59
60	static av_cold int audio_read_header(AVFormatContext *s1)
61	{
62	AlsaData *s = s1->priv_data;
63	AVStream *st;
64	int ret;
65	enum AVCodecID codec_id;
66
67	st = avformat_new_stream(s1, NULL);
68	if (!st) {
69	av_log(s1, AV_LOG_ERROR, "Cannot add stream\n");
70
71	return AVERROR(ENOMEM);
72	}
73	codec_id = s1->audio_codec_id;
74
75	ret = ff_alsa_open(s1, SND_PCM_STREAM_CAPTURE, &s->sample_rate, s->channels,
76	&codec_id);
77	if (ret < 0) {
78	return AVERROR(EIO);
79	}
80
81	/* take real parameters */
82	st->codecpar->codec_type = AVMEDIA_TYPE_AUDIO;
83	st->codecpar->codec_id = codec_id;
84	st->codecpar->sample_rate = s->sample_rate;
85	st->codecpar->channels = s->channels;
86	st->codecpar->frame_size = s->frame_size;
87	avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
88	/* microseconds instead of seconds, MHz instead of Hz */
89	s->timefilter = ff_timefilter_new(1000000.0 / s->sample_rate,
90	s->period_size, 1.5E-6);
91	if (!s->timefilter)
92	goto fail;
93
94	return 0;
95
96	fail:
97	snd_pcm_close(s->h);
98	return AVERROR(EIO);
99	}
100
101	static int audio_read_packet(AVFormatContext *s1, AVPacket *pkt)
102	{
103	AlsaData *s = s1->priv_data;
104	int res;
105	int64_t dts;
106	snd_pcm_sframes_t delay = 0;
107
108	if (av_new_packet(pkt, s->period_size * s->frame_size) < 0) {
109	return AVERROR(EIO);
110	}
111
112	while ((res = snd_pcm_readi(s->h, pkt->data, s->period_size)) < 0) {
113	if (res == -EAGAIN) {
114	av_packet_unref(pkt);
115
116	return AVERROR(EAGAIN);
117	}
118	if (ff_alsa_xrun_recover(s1, res) < 0) {
119	av_log(s1, AV_LOG_ERROR, "ALSA read error: %s\n",
120	snd_strerror(res));
121	av_packet_unref(pkt);
122
123	return AVERROR(EIO);
124	}
125	ff_timefilter_reset(s->timefilter);
126	}
127
128	dts = av_gettime();
129	snd_pcm_delay(s->h, &delay);
130	dts -= av_rescale(delay + res, 1000000, s->sample_rate);
131	pkt->pts = ff_timefilter_update(s->timefilter, dts, s->last_period);
132	s->last_period = res;
133
134	pkt->size = res * s->frame_size;
135
136	return 0;
137	}
138
139	static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
140	{
141	return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_CAPTURE);
142	}
143
144	static const AVOption options[] = {
145	{ "sample_rate", "", offsetof(AlsaData, sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
146	{ "channels", "", offsetof(AlsaData, channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, AV_OPT_FLAG_DECODING_PARAM },
147	{ NULL },
148	};
149
150	static const AVClass alsa_demuxer_class = {
151	.class_name = "ALSA demuxer",
152	.item_name = av_default_item_name,
153	.option = options,
154	.version = LIBAVUTIL_VERSION_INT,
155	.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
156	};
157
158	AVInputFormat ff_alsa_demuxer = {
159	.name = "alsa",
160	.long_name = NULL_IF_CONFIG_SMALL("ALSA audio input"),
161	.priv_data_size = sizeof(AlsaData),
162	.read_header = audio_read_header,
163	.read_packet = audio_read_packet,
164	.read_close = ff_alsa_close,
165	.get_device_list = audio_get_device_list,
166	.flags = AVFMT_NOFILE,
167	.priv_class = &alsa_demuxer_class,
168	};
169