1
2
3
4
5
6
7
8
9
10
11
12
13
14
15
16
17
18
19
20
21
22
23
24
25
26
27
28
29
30
31
32
33
34
35
36
37
38
39
40
41
42
43
44
45
46
47
48
49
50
51
52
53
54
55
56
57
58
59
60
61
62
63
64
65
66
67
68
69
70
71
72
73
74
75
76
77
78
79
80
81
82
83
84
85
86
87
88
89
90
91
92
93
94
95
96
97
98
99
100
101
102
103
104
105
106
107
108
109
110
111
112
113
114
115
116
117
118
119
120
121
122
123
124
125
126
127
128
129
130
131
132
133
134
135
136
137
138
139
140
141
142
143
144
145
146
147
148
149
150
151
152
153
154
155
156
157
158
159
160
161
162
163
164
165
166
167
168
169
170
171
172
173
174
175
176
177
178
179
180
181
182
183
184
185
186
187
188
189
190
|
/*
* Pulseaudio input
* Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
*
* This file is part of Libav.
*
* Libav is free software; you can redistribute it and/or
* modify it under the terms of the GNU Lesser General Public
* License as published by the Free Software Foundation; either
* version 2.1 of the License, or (at your option) any later version.
*
* Libav is distributed in the hope that it will be useful,
* but WITHOUT ANY WARRANTY; without even the implied warranty of
* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
* Lesser General Public License for more details.
*
* You should have received a copy of the GNU Lesser General Public
* License along with Libav; if not, write to the Free Software
* Foundation, Inc., 51 Franklin Street, Fifth Floor, Boston, MA 02110-1301 USA
*/
/**
* @file
* PulseAudio input using the simple API.
* @author Luca Barbato <lu_zero@gentoo.org>
*/
#include <pulse/simple.h>
#include <pulse/rtclock.h>
#include <pulse/error.h>
#include "libavformat/avformat.h"
#include "libavformat/internal.h"
#include "libavutil/opt.h"
#define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
typedef struct PulseData {
AVClass *class;
char *server;
char *name;
char *stream_name;
int sample_rate;
int channels;
int frame_size;
int fragment_size;
pa_simple *s;
int64_t pts;
int64_t frame_duration;
} PulseData;
static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
switch (codec_id) {
case AV_CODEC_ID_PCM_U8: return PA_SAMPLE_U8;
case AV_CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW;
case AV_CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
case AV_CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
case AV_CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
case AV_CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
case AV_CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
case AV_CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
case AV_CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
case AV_CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
case AV_CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
default: return PA_SAMPLE_INVALID;
}
}
static av_cold int pulse_read_header(AVFormatContext *s)
{
PulseData *pd = s->priv_data;
AVStream *st;
char *device = NULL;
int ret;
enum AVCodecID codec_id =
s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
const pa_sample_spec ss = { codec_id_to_pulse_format(codec_id),
pd->sample_rate,
pd->channels };
pa_buffer_attr attr = { -1 };
st = avformat_new_stream(s, NULL);
if (!st) {
av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
return AVERROR(ENOMEM);
}
attr.fragsize = pd->fragment_size;
if (strcmp(s->filename, "default"))
device = s->filename;
pd->s = pa_simple_new(pd->server, pd->name,
PA_STREAM_RECORD,
device, pd->stream_name, &ss,
NULL, &attr, &ret);
if (!pd->s) {
av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
pa_strerror(ret));
return AVERROR(EIO);
}
/* take real parameters */
st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
st->codec->codec_id = codec_id;
st->codec->sample_rate = pd->sample_rate;
st->codec->channels = pd->channels;
avpriv_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
pd->pts = AV_NOPTS_VALUE;
pd->frame_duration = (pd->frame_size * 1000000LL * 8) /
(pd->sample_rate * pd->channels * av_get_bits_per_sample(codec_id));
return 0;
}
static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
{
PulseData *pd = s->priv_data;
int res;
pa_usec_t latency;
if (av_new_packet(pkt, pd->frame_size) < 0) {
return AVERROR(ENOMEM);
}
if ((pa_simple_read(pd->s, pkt->data, pkt->size, &res)) < 0) {
av_log(s, AV_LOG_ERROR, "pa_simple_read failed: %s\n",
pa_strerror(res));
av_free_packet(pkt);
return AVERROR(EIO);
}
if ((latency = pa_simple_get_latency(pd->s, &res)) == (pa_usec_t) -1) {
av_log(s, AV_LOG_ERROR, "pa_simple_get_latency() failed: %s\n",
pa_strerror(res));
return AVERROR(EIO);
}
if (pd->pts == AV_NOPTS_VALUE) {
pd->pts = -latency;
}
pkt->pts = pd->pts;
pd->pts += pd->frame_duration;
return 0;
}
static av_cold int pulse_close(AVFormatContext *s)
{
PulseData *pd = s->priv_data;
pa_simple_free(pd->s);
return 0;
}
#define OFFSET(a) offsetof(PulseData, a)
#define D AV_OPT_FLAG_DECODING_PARAM
static const AVOption options[] = {
{ "server", "pulse server name", OFFSET(server), AV_OPT_TYPE_STRING, {.str = NULL}, 0, 0, D },
{ "name", "application name", OFFSET(name), AV_OPT_TYPE_STRING, {.str = LIBAVFORMAT_IDENT}, 0, 0, D },
{ "stream_name", "stream description", OFFSET(stream_name), AV_OPT_TYPE_STRING, {.str = "record"}, 0, 0, D },
{ "sample_rate", "sample rate in Hz", OFFSET(sample_rate), AV_OPT_TYPE_INT, {.i64 = 48000}, 1, INT_MAX, D },
{ "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.i64 = 2}, 1, INT_MAX, D },
{ "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.i64 = 1024}, 1, INT_MAX, D },
{ "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.i64 = -1}, -1, INT_MAX, D },
{ NULL },
};
static const AVClass pulse_demuxer_class = {
.class_name = "Pulse demuxer",
.item_name = av_default_item_name,
.option = options,
.version = LIBAVUTIL_VERSION_INT,
};
AVInputFormat ff_pulse_demuxer = {
.name = "pulse",
.long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
.priv_data_size = sizeof(PulseData),
.read_header = pulse_read_header,
.read_packet = pulse_read_packet,
.read_close = pulse_close,
.flags = AVFMT_NOFILE,
.priv_class = &pulse_demuxer_class,
};
|
