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
audio_write_header(AVFormatContext * s1)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
audio_write_packet(AVFormatContext * s1,AVPacket * pkt)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
audio_write_frame(AVFormatContext * s1,int stream_index,AVFrame ** frame,unsigned flags)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 = (*frame)->pkt_duration;
134 return audio_write_packet(s1, &pkt);
135 }
136
137 static void
audio_get_output_timestamp(AVFormatContext * s1,int stream,int64_t * dts,int64_t * wall)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
audio_get_device_list(AVFormatContext * h,AVDeviceInfoList * device_list)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 outdev",
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