pulse: introduce pulseaudio input

It currently use the simple api and is using the latency information
provided only to offset the stream start.

Signed-off-by: Luca Barbato <lu_zero@gentoo.org>

1 files changed, 190 insertions, 0 deletions

diff --git a/libavdevice/pulse.c b/libavdevice/pulse.c
new file mode 100644
index 0000000000..1edd24fdd9
--- /dev/null
+++ b/libavdevice/pulse.c
@@ -0,0 +1,190 @@
+/*
+ * 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 "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,
+};