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pulse_audio_dec.c
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1 /*
2  * Pulseaudio input
3  * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
4  *
5  * This file is part of FFmpeg.
6  *
7  * FFmpeg is free software; you can redistribute it and/or
8  * modify it under the terms of the GNU Lesser General Public
9  * License as published by the Free Software Foundation; either
10  * version 2.1 of the License, or (at your option) any later version.
11  *
12  * FFmpeg is distributed in the hope that it will be useful,
13  * but WITHOUT ANY WARRANTY; without even the implied warranty of
14  * MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
15  * Lesser General Public License for more details.
16  *
17  * You should have received a copy of the GNU Lesser General Public
18  * License along with FFmpeg; if not, write to the Free Software
19  * Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
20  */
21 
22 /**
23  * @file
24  * PulseAudio input using the simple API.
25  * @author Luca Barbato <lu_zero@gentoo.org>
26  */
27 
28 #include <pulse/simple.h>
29 #include <pulse/rtclock.h>
30 #include <pulse/error.h>
31 #include "libavformat/avformat.h"
32 #include "libavformat/internal.h"
33 #include "libavutil/opt.h"
34 #include "pulse_audio_common.h"
35 
36 #define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
37 
38 typedef struct PulseData {
39  AVClass *class;
40  char *server;
41  char *name;
42  char *stream_name;
44  int channels;
47  pa_simple *s;
48  int64_t pts;
49  int64_t frame_duration;
50 } PulseData;
51 
53 {
54  PulseData *pd = s->priv_data;
55  AVStream *st;
56  char *device = NULL;
57  int ret, sample_bytes;
58  enum AVCodecID codec_id =
60  const pa_sample_spec ss = { ff_codec_id_to_pulse_format(codec_id),
61  pd->sample_rate,
62  pd->channels };
63 
64  pa_buffer_attr attr = { -1 };
65 
66  st = avformat_new_stream(s, NULL);
67 
68  if (!st) {
69  av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
70  return AVERROR(ENOMEM);
71  }
72 
73  attr.fragsize = pd->fragment_size;
74 
75  if (strcmp(s->filename, "default"))
76  device = s->filename;
77 
78  pd->s = pa_simple_new(pd->server, pd->name,
79  PA_STREAM_RECORD,
80  device, pd->stream_name, &ss,
81  NULL, &attr, &ret);
82 
83  if (!pd->s) {
84  av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
85  pa_strerror(ret));
86  return AVERROR(EIO);
87  }
88  /* take real parameters */
90  st->codec->codec_id = codec_id;
91  st->codec->sample_rate = pd->sample_rate;
92  st->codec->channels = pd->channels;
93  avpriv_set_pts_info(st, 64, 1, pd->sample_rate); /* 64 bits pts in us */
94 
95  pd->pts = AV_NOPTS_VALUE;
96  sample_bytes = (av_get_bits_per_sample(codec_id) >> 3) * pd->channels;
97 
98  if (pd->frame_size % sample_bytes) {
99  av_log(s, AV_LOG_WARNING, "frame_size %i is not divisible by %i "
100  "(channels * bytes_per_sample) \n", pd->frame_size, sample_bytes);
101  }
102 
103  pd->frame_duration = pd->frame_size / sample_bytes;
104 
105  return 0;
106 }
107 
109 {
110  PulseData *pd = s->priv_data;
111  int res;
112 
113  if (av_new_packet(pkt, pd->frame_size) < 0) {
114  return AVERROR(ENOMEM);
115  }
116 
117  if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
118  av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
119  pa_strerror(res));
120  av_free_packet(pkt);
121  return AVERROR(EIO);
122  }
123 
124  if (pd->pts == AV_NOPTS_VALUE) {
125  pa_usec_t latency;
126 
127  if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
128  av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
129  pa_strerror(res));
130  return AVERROR(EIO);
131  }
132 
133  pd->pts = -latency;
134  }
135 
136  pkt->pts = pd->pts;
137 
138  pd->pts += pd->frame_duration;
139 
140  return 0;
141 }
142 
144 {
145  PulseData *pd = s->priv_data;
146  pa_simple_free(pd->s);
147  return 0;
148 }
149 
150 #define OFFSET(a) offsetof(PulseData, a)
151 #define D AV_OPT_FLAG_DECODING_PARAM
152 
153 static const AVOption options[] = {
154  { "server", "set PulseAudio server", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
155  { "name", "set application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
156  { "stream_name", "set stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
157  { "sample_rate", "set sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D },
158  { "channels", "set number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D },
159  { "frame_size", "set number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D },
160  { "fragment_size", "set buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D },
161  { NULL },
162 };
163 
164 static const AVClass pulse_demuxer_class = {
165  .class_name = "Pulse demuxer",
166  .item_name = av_default_item_name,
167  .option = options,
168  .version = LIBAVUTIL_VERSION_INT,
169 };
170 
172  .name = "pulse",
173  .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
174  .priv_data_size = sizeof(PulseData),
178  .flags = AVFMT_NOFILE,
179  .priv_class = &pulse_demuxer_class,
180 };