--- /dev/null
+/*
+ * 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 "libavutil/opt.h"
+
+#define DEFAULT_CODEC_ID AV_NE(CODEC_ID_PCM_S16BE, 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;
+} PulseData;
+
+static pa_sample_format_t codec_id_to_pulse_format(int codec_id) {
+ switch (codec_id) {
+ case CODEC_ID_PCM_U8: return PA_SAMPLE_U8;
+ case CODEC_ID_PCM_ALAW: return PA_SAMPLE_ALAW;
+ case CODEC_ID_PCM_MULAW: return PA_SAMPLE_ULAW;
+ case CODEC_ID_PCM_S16LE: return PA_SAMPLE_S16LE;
+ case CODEC_ID_PCM_S16BE: return PA_SAMPLE_S16BE;
+ case CODEC_ID_PCM_F32LE: return PA_SAMPLE_FLOAT32LE;
+ case CODEC_ID_PCM_F32BE: return PA_SAMPLE_FLOAT32BE;
+ case CODEC_ID_PCM_S32LE: return PA_SAMPLE_S32LE;
+ case CODEC_ID_PCM_S32BE: return PA_SAMPLE_S32BE;
+ case CODEC_ID_PCM_S24LE: return PA_SAMPLE_S24LE;
+ case CODEC_ID_PCM_S24BE: return PA_SAMPLE_S24BE;
+ default: return PA_SAMPLE_INVALID;
+ }
+}
+
+static av_cold int pulse_read_header(AVFormatContext *s,
+ AVFormatParameters *ap)
+{
+ PulseData *pd = s->priv_data;
+ AVStream *st;
+ char *device = NULL;
+ int ret;
+ enum CodecID codec_id =
+ s->audio_codec_id == 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;
+ av_set_pts_info(st, 64, 1, 1000000); /* 64 bits pts in us */
+
+ pd->pts = AV_NOPTS_VALUE;
+
+ return 0;
+}
+
+static int pulse_read_packet(AVFormatContext *s, AVPacket *pkt)
+{
+ PulseData *pd = s->priv_data;
+ int res;
+ pa_usec_t latency;
+ uint64_t frame_duration =
+ (pd->frame_size*1000000LL) / (pd->sample_rate * pd->channels);
+
+ 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 += 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 = "libav"}, 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, {.dbl = 48000}, 1, INT_MAX, D },
+ { "channels", "number of audio channels", OFFSET(channels), AV_OPT_TYPE_INT, {.dbl = 2}, 1, INT_MAX, D },
+ { "frame_size", "number of bytes per frame", OFFSET(frame_size), AV_OPT_TYPE_INT, {.dbl = 1024}, 1, INT_MAX, D },
+ { "fragment_size", "buffering size, affects latency and cpu usage", OFFSET(fragment_size), AV_OPT_TYPE_INT, {.dbl = -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,
+};