3 * Copyright (c) 2011 Luca Barbato <lu_zero@gentoo.org>
5 * This file is part of FFmpeg.
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.
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.
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
24 * PulseAudio input using the simple API.
25 * @author Luca Barbato <lu_zero@gentoo.org>
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"
36 #define DEFAULT_CODEC_ID AV_NE(AV_CODEC_ID_PCM_S16BE, AV_CODEC_ID_PCM_S16LE)
38 typedef struct PulseData {
49 int64_t frame_duration;
52 static av_cold int pulse_read_header(AVFormatContext *s)
54 PulseData *pd = s->priv_data;
57 int ret, sample_bytes;
58 enum AVCodecID codec_id =
59 s->audio_codec_id == AV_CODEC_ID_NONE ? DEFAULT_CODEC_ID : s->audio_codec_id;
60 const pa_sample_spec ss = { ff_codec_id_to_pulse_format(codec_id),
64 pa_buffer_attr attr = { -1 };
66 st = avformat_new_stream(s, NULL);
69 av_log(s, AV_LOG_ERROR, "Cannot add stream\n");
70 return AVERROR(ENOMEM);
73 attr.fragsize = pd->fragment_size;
75 if (strcmp(s->filename, "default"))
78 pd->s = pa_simple_new(pd->server, pd->name,
80 device, pd->stream_name, &ss,
84 av_log(s, AV_LOG_ERROR, "pa_simple_new failed: %s\n",
88 /* take real parameters */
89 st->codec->codec_type = AVMEDIA_TYPE_AUDIO;
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 */
95 pd->pts = AV_NOPTS_VALUE;
96 sample_bytes = (av_get_bits_per_sample(codec_id) >> 3) * pd->channels;
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);
103 pd->frame_duration = pd->frame_size / sample_bytes;
108 static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
110 PulseData *pd = s->priv_data;
113 if (av_new_packet(pkt, pd->frame_size) < 0) {
114 return AVERROR(ENOMEM);
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",
124 if (pd->pts == AV_NOPTS_VALUE) {
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",
138 pd->pts += pd->frame_duration;
143 static av_cold int pulse_close(AVFormatContext *s)
145 PulseData *pd = s->priv_data;
146 pa_simple_free(pd->s);
150 #define OFFSET(a) offsetof(PulseData, a)
151 #define D AV_OPT_FLAG_DECODING_PARAM
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 },
164 static const AVClass pulse_demuxer_class = {
165 .class_name = "Pulse demuxer",
166 .item_name = av_default_item_name,
168 .version = LIBAVUTIL_VERSION_INT,
169 .category = AV_CLASS_CATEGORY_DEVICE_AUDIO_INPUT,
172 AVInputFormat ff_pulse_demuxer = {
174 .long_name = NULL_IF_CONFIG_SMALL("Pulse audio input"),
175 .priv_data_size = sizeof(PulseData),
176 .read_header = pulse_read_header,
177 .read_packet = pulse_read_packet,
178 .read_close = pulse_close,
179 .flags = AVFMT_NOFILE,
180 .priv_class = &pulse_demuxer_class,