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pulse.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 Libav.
6  *
7  * Libav 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  * Libav 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 Libav; 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 
32 #include "libavformat/avformat.h"
33 #include "libavformat/internal.h"
34 #include "libavutil/opt.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;
43  int sample_rate;
44  int channels;
45  int frame_size;
46  int fragment_size;
47  pa_simple *s;
48  int64_t pts;
49  int64_t frame_duration;
50 } PulseData;
51 
52 static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
53  switch (codec_id) {
54  case AV_CODEC_ID_PCM_U8: return PA_SAMPLE_U8;
55  case AV_CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW;
56  case AV_CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
57  case AV_CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
58  case AV_CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
59  case AV_CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
60  case AV_CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
61  case AV_CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
62  case AV_CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
63  case AV_CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
64  case AV_CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
65  default: return PA_SAMPLE_INVALID;
66  }
67 }
68 
69 static av_cold int pulse_read_header(AVFormatContext *s)
70 {
71  PulseData *pd = s->priv_data;
72  AVStream *st;
73  char *device = NULL;
74  int ret;
75  enum AVCodecID codec_id =
76  s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
77  const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
78  pd->sample_rate,
79  pd->channels };
80 
81  pa_buffer_attr attr = { -1 };
82 
83  st = avformat_new_stream(s, NULL);
84 
85  if (!st) {
86  av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
87  return AVERROR(ENOMEM);
88  }
89 
90  attr.fragsize = pd->fragment_size;
91 
92  if (strcmp(s->filename, "default"))
93  device = s->filename;
94 
95  pd->s = pa_simple_new(pd->server, pd->name,
96  PA_STREAM_RECORD,
97  device, pd->stream_name, &ss,
98  NULL, &attr, &ret);
99 
100  if (!pd->s) {
101  av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
102  pa_strerror(ret));
103  return AVERROR(EIO);
104  }
105  /* take real parameters */
106  st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
107  st->codec->codec_id = codec_id;
108  st->codec->sample_rate = pd->sample_rate;
109  st->codec->channels = pd->channels;
110  avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
111 
112  pd->pts = AV_NOPTS_VALUE;
113  pd->frame_duration = (pd->frame_size * 1000000LL * 8) /
114  (pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));
115 
116  return 0;
117 }
118 
119 static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
120 {
121  PulseData *pd = s->priv_data;
122  int res;
123  pa_usec_t latency;
124 
125  if (av_new_packet(pkt, pd->frame_size) < 0) {
126  return AVERROR(ENOMEM);
127  }
128 
129  if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
130  av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
131  pa_strerror(res));
132  av_free_packet(pkt);
133  return AVERROR(EIO);
134  }
135 
136  if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
137  av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
138  pa_strerror(res));
139  return AVERROR(EIO);
140  }
141 
142  if (pd->pts == AV_NOPTS_VALUE) {
143  pd->pts = -latency;
144  }
145 
146  pkt->pts = pd->pts;
147 
148  pd->pts += pd->frame_duration;
149 
150  return 0;
151 }
152 
153 static av_cold int pulse_close(AVFormatContext *s)
154 {
155  PulseData *pd = s->priv_data;
156  pa_simple_free(pd->s);
157  return 0;
158 }
159 
160 #define OFFSET(a) offsetof(PulseData, a)
161 #define D AV_OPT_FLAG_DECODING_PARAM
162 
163 static const AVOption options[] = {
164  { "server", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
165  { "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
166  { "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
167  { "sample_rate", "sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D },
168  { "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D },
169  { "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D },
170  { "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D },
171  { NULL },
172 };
173 
174 static const AVClass pulse_demuxer_class = {
175  .class_name = "Pulse demuxer",
176  .item_name = av_default_item_name,
177  .option = options,
178  .version = LIBAVUTIL_VERSION_INT,
179 };
180 
181 AVInputFormat ff_pulse_demuxer = {
182  .name = "pulse",
183  .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
184  .priv_data_size = sizeof(PulseData),
185  .read_header = pulse_read_header,
186  .read_packet = pulse_read_packet,
187  .read_close = pulse_close,
188  .flags = AVFMT_NOFILE,
189  .priv_class = &pulse_demuxer_class,
190 };
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