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[libav.git] / libavdevice / 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>
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26 */
27
28#include <pulse/simple.h>
29#include <pulse/rtclock.h>
30#include <pulse/error.h>
31
32#include "libavformat/avformat.h"
c3f9ebf7 33#include "libavformat/internal.h"
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34#include "libavutil/opt.h"
35
36#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, CODEC_ID_PCM_S16LE)
37
38typedef 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;
2625b621 49 int64_t frame_duration;
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50} PulseData;
51
52static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
53 switch (codec_id) {
54 case CODEC_ID_PCM_U8: return PA_SAMPLE_U8;
55 case CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW;
56 case CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
57 case CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
58 case CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
59 case CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
60 case CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
61 case CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
62 case CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
63 case CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
64 case CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
65 default: return PA_SAMPLE_INVALID;
66 }
67}
68
69static av_cold int pulse_read_header(AVFormatContext *s,
70 AVFormatParameters *ap)
71{
72 PulseData *pd = s->priv_data;
73 AVStream *st;
74 char *device = NULL;
75 int ret;
76 enum CodecID codec_id =
77 s->audio_codec_id == CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
78 const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
79 pd->sample_rate,
80 pd->channels };
81
82 pa_buffer_attr attr = { -1 };
83
84 st = avformat_new_stream(s, NULL);
85
86 if (!st) {
87 av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
88 return AVERROR(ENOMEM);
89 }
90
91 attr.fragsize = pd->fragment_size;
92
93 if (strcmp(s->filename, "default"))
94 device = s->filename;
95
96 pd->s = pa_simple_new(pd->server, pd->name,
97 PA_STREAM_RECORD,
98 device, pd->stream_name, &ss,
99 NULL, &attr, &ret);
100
101 if (!pd->s) {
102 av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
103 pa_strerror(ret));
104 return AVERROR(EIO);
105 }
106 /* take real parameters */
107 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
108 st->codec->codec_id = codec_id;
109 st->codec->sample_rate = pd->sample_rate;
110 st->codec->channels = pd->channels;
c3f9ebf7 111 avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
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112
113 pd->pts = AV_NOPTS_VALUE;
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114 pd->frame_duration = (pd->frame_size * 1000000LL * 8) /
115 (pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));
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116
117 return 0;
118}
119
120static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
121{
122 PulseData *pd = s->priv_data;
123 int res;
124 pa_usec_t latency;
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125
126 if (av_new_packet(pkt, pd->frame_size) < 0) {
127 return AVERROR(ENOMEM);
128 }
129
130 if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
131 av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
132 pa_strerror(res));
133 av_free_packet(pkt);
134 return AVERROR(EIO);
135 }
136
137 if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
138 av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
139 pa_strerror(res));
140 return AVERROR(EIO);
141 }
142
143 if (pd->pts == AV_NOPTS_VALUE) {
144 pd->pts = -latency;
145 }
146
147 pkt->pts = pd->pts;
148
2625b621 149 pd->pts += pd->frame_duration;
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150
151 return 0;
152}
153
154static av_cold int pulse_close(AVFormatContext *s)
155{
156 PulseData *pd = s->priv_data;
157 pa_simple_free(pd->s);
158 return 0;
159}
160
161#define OFFSET(a) offsetof(PulseData, a)
162#define D AV_OPT_FLAG_DECODING_PARAM
163
164static const AVOption options[] = {
165 { "server", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
166 { "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = "libav"}, 0, 0, D },
167 { "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
168 { "sample_rate", "sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.dbl = 48000}, 1, INT_MAX, D },
169 { "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D },
170 { "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 1, INT_MAX, D },
171 { "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.dbl = -1}, -1, INT_MAX, D },
172 { NULL },
173};
174
175static const AVClass pulse_demuxer_class = {
176 .class_name = "Pulse demuxer",
177 .item_name = av_default_item_name,
178 .option = options,
179 .version = LIBAVUTIL_VERSION_INT,
180};
181
182AVInputFormat ff_pulse_demuxer = {
183 .name = "pulse",
184 .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
185 .priv_data_size = sizeof(PulseData),
186 .read_header = pulse_read_header,
187 .read_packet = pulse_read_packet,
188 .read_close = pulse_close,
189 .flags = AVFMT_NOFILE,
190 .priv_class = &pulse_demuxer_class,
191};