FFmpeg: libavdevice/alsa-audio-enc.c Source File

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