path: root/libavdevice/alsa_enc.c (plain)
blob: bddc61f4aaf4c563329b04c4e53ae2781b81dcdb

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: output
26	* @author Luca Abeni ( lucabe72 email it )
27	* @author Benoit Fouet ( benoit fouet free fr )
28	*
29	* This avdevice encoder can 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	#include "libavutil/time.h"
44
45
46	#include "libavformat/internal.h"
47	#include "avdevice.h"
48	#include "alsa.h"
49
50	static av_cold int audio_write_header(AVFormatContext *s1)
51	{
52	AlsaData *s = s1->priv_data;
53	AVStream *st = NULL;
54	unsigned int sample_rate;
55	enum AVCodecID codec_id;
56	int res;
57
58	if (s1->nb_streams != 1 || s1->streams[0]->codecpar->codec_type != AVMEDIA_TYPE_AUDIO) {
59	av_log(s1, AV_LOG_ERROR, "Only a single audio stream is supported.\n");
60	return AVERROR(EINVAL);
61	}
62	st = s1->streams[0];
63
64	sample_rate = st->codecpar->sample_rate;
65	codec_id = st->codecpar->codec_id;
66	res = ff_alsa_open(s1, SND_PCM_STREAM_PLAYBACK, &sample_rate,
67	st->codecpar->channels, &codec_id);
68	if (sample_rate != st->codecpar->sample_rate) {
69	av_log(s1, AV_LOG_ERROR,
70	"sample rate %d not available, nearest is %d\n",
71	st->codecpar->sample_rate, sample_rate);
72	goto fail;
73	}
74	avpriv_set_pts_info(st, 64, 1, sample_rate);
75
76	return res;
77
78	fail:
79	snd_pcm_close(s->h);
80	return AVERROR(EIO);
81	}
82
83	static int audio_write_packet(AVFormatContext *s1, AVPacket *pkt)
84	{
85	AlsaData *s = s1->priv_data;
86	int res;
87	int size = pkt->size;
88	uint8_t *buf = pkt->data;
89
90	size /= s->frame_size;
91	if (pkt->dts != AV_NOPTS_VALUE)
92	s->timestamp = pkt->dts;
93	s->timestamp += pkt->duration ? pkt->duration : size;
94
95	if (s->reorder_func) {
96	if (size > s->reorder_buf_size)
97	if (ff_alsa_extend_reorder_buf(s, size))
98	return AVERROR(ENOMEM);
99	s->reorder_func(buf, s->reorder_buf, size);
100	buf = s->reorder_buf;
101	}
102	while ((res = snd_pcm_writei(s->h, buf, size)) < 0) {
103	if (res == -EAGAIN) {
104
105	return AVERROR(EAGAIN);
106	}
107
108	if (ff_alsa_xrun_recover(s1, res) < 0) {
109	av_log(s1, AV_LOG_ERROR, "ALSA write error: %s\n",
110	snd_strerror(res));
111
112	return AVERROR(EIO);
113	}
114	}
115
116	return 0;
117	}
118
119	static int audio_write_frame(AVFormatContext *s1, int stream_index,
120	AVFrame **frame, unsigned flags)
121	{
122	AlsaData *s = s1->priv_data;
123	AVPacket pkt;
124
125	/* ff_alsa_open() should have accepted only supported formats */
126	if ((flags & AV_WRITE_UNCODED_FRAME_QUERY))
127	return av_sample_fmt_is_planar(s1->streams[stream_index]->codecpar->format) ?
128	AVERROR(EINVAL) : 0;
129	/* set only used fields */
130	pkt.data = (*frame)->data[0];
131	pkt.size = (*frame)->nb_samples * s->frame_size;
132	pkt.dts = (*frame)->pkt_dts;
133	pkt.duration = av_frame_get_pkt_duration(*frame);
134	return audio_write_packet(s1, &pkt);
135	}
136
137	static void
138	audio_get_output_timestamp(AVFormatContext *s1, int stream,
139	int64_t *dts, int64_t *wall)
140	{
141	AlsaData *s = s1->priv_data;
142	snd_pcm_sframes_t delay = 0;
143	*wall = av_gettime();
144	snd_pcm_delay(s->h, &delay);
145	*dts = s->timestamp - delay;
146	}
147
148	static int audio_get_device_list(AVFormatContext *h, AVDeviceInfoList *device_list)
149	{
150	return ff_alsa_get_device_list(device_list, SND_PCM_STREAM_PLAYBACK);
151	}
152
153	static const AVClass alsa_muxer_class = {
154	.class_name = "ALSA muxer",
155	.item_name = av_default_item_name,
156	.version = LIBAVUTIL_VERSION_INT,
157	.category = AV_CLASS_CATEGORY_DEVICE_AUDIO_OUTPUT,
158	};
159
160	AVOutputFormat ff_alsa_muxer = {
161	.name = "alsa",
162	.long_name = NULL_IF_CONFIG_SMALL("ALSA audio output"),
163	.priv_data_size = sizeof(AlsaData),
164	.audio_codec = DEFAULT_CODEC_ID,
165	.video_codec = AV_CODEC_ID_NONE,
166	.write_header = audio_write_header,
167	.write_packet = audio_write_packet,
168	.write_trailer = ff_alsa_close,
169	.write_uncoded_frame = audio_write_frame,
170	.get_device_list = audio_get_device_list,
171	.get_output_timestamp = audio_get_output_timestamp,
172	.flags = AVFMT_NOFILE,
173	.priv_class = &alsa_muxer_class,
174	};
175