gstreamer/ext/srt/gstsrtclientsrc.c
Justin Kim 1da40cdc0f srt: Remove platform dependent socket headers
SRT modules entrust `gnetworking.h` with finding right headers
for the platforms.

https://bugzilla.gnome.org/show_bug.cgi?id=792123
2018-01-03 10:41:45 +00:00

337 lines
10 KiB
C

/* GStreamer SRT plugin based on libsrt
* Copyright (C) 2017, Collabora Ltd.
* Author:Justin Kim <justin.kim@collabora.com>
*
* This library is free software; you can redistribute it and/or
* modify it under the terms of the GNU Library General Public
* License as published by the Free Software Foundation; either
* version 2 of the License, or (at your option) any later version.
*
* This library 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
* Library General Public License for more details.
*
* You should have received a copy of the GNU Library General Public
* License along with this library; if not, write to the
* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
* Boston, MA 02110-1301, USA.
*/
/**
* SECTION:element-srtclientsrc
* @title: srtclientsrc
*
* srtclientsrc is a network source that reads <ulink url="http://www.srtalliance.org/">SRT</ulink>
* packets from the network. Although SRT is a protocol based on UDP, srtclientsrc works like
* a client socket of connection-oriented protocol.
*
* <refsect2>
* <title>Examples</title>
* |[
* gst-launch-1.0 -v srtclientsrc uri="srt://127.0.0.1:7001" ! fakesink
* ]| This pipeline shows how to connect SRT server by setting #GstSRTClientSrc:uri property.
*
* |[
* gst-launch-1.0 -v srtclientsrc uri="srt://192.168.1.10:7001" rendez-vous ! fakesink
* ]| This pipeline shows how to connect SRT server by setting #GstSRTClientSrc:uri property and using the rendez-vous mode.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include "config.h"
#endif
#include "gstsrtclientsrc.h"
#include <srt/srt.h>
#include <gio/gio.h>
#include "gstsrt.h"
static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src",
GST_PAD_SRC,
GST_PAD_ALWAYS,
GST_STATIC_CAPS_ANY);
#define GST_CAT_DEFAULT gst_debug_srt_client_src
GST_DEBUG_CATEGORY (GST_CAT_DEFAULT);
struct _GstSRTClientSrcPrivate
{
SRTSOCKET sock;
gint poll_id;
gint poll_timeout;
gboolean rendez_vous;
gchar *bind_address;
guint16 bind_port;
};
#define GST_SRT_CLIENT_SRC_GET_PRIVATE(obj) \
(G_TYPE_INSTANCE_GET_PRIVATE ((obj), GST_TYPE_SRT_CLIENT_SRC, GstSRTClientSrcPrivate))
#define SRT_DEFAULT_POLL_TIMEOUT -1
enum
{
PROP_POLL_TIMEOUT = 1,
PROP_BIND_ADDRESS,
PROP_BIND_PORT,
PROP_RENDEZ_VOUS,
/*< private > */
PROP_LAST
};
static GParamSpec *properties[PROP_LAST + 1];
#define gst_srt_client_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstSRTClientSrc, gst_srt_client_src,
GST_TYPE_SRT_BASE_SRC, G_ADD_PRIVATE (GstSRTClientSrc)
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "srtclientsrc", 0,
"SRT Client Source"));
static void
gst_srt_client_src_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (object);
GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
switch (prop_id) {
case PROP_POLL_TIMEOUT:
g_value_set_int (value, priv->poll_timeout);
break;
case PROP_BIND_PORT:
g_value_set_int (value, priv->rendez_vous);
break;
case PROP_BIND_ADDRESS:
g_value_set_string (value, priv->bind_address);
break;
case PROP_RENDEZ_VOUS:
g_value_set_boolean (value, priv->bind_port);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_srt_client_src_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstSRTBaseSrc *self = GST_SRT_BASE_SRC (object);
GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
switch (prop_id) {
case PROP_POLL_TIMEOUT:
priv->poll_timeout = g_value_get_int (value);
break;
case PROP_BIND_ADDRESS:
g_free (priv->bind_address);
priv->bind_address = g_value_dup_string (value);
break;
case PROP_BIND_PORT:
priv->bind_port = g_value_get_int (value);
break;
case PROP_RENDEZ_VOUS:
priv->rendez_vous = g_value_get_boolean (value);
break;
default:
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
break;
}
}
static void
gst_srt_client_src_finalize (GObject * object)
{
GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (object);
GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
if (priv->poll_id != SRT_ERROR) {
srt_epoll_release (priv->poll_id);
priv->poll_id = SRT_ERROR;
}
if (priv->sock != SRT_INVALID_SOCK) {
srt_close (priv->sock);
priv->sock = SRT_INVALID_SOCK;
}
g_free (priv->bind_address);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static GstFlowReturn
gst_srt_client_src_fill (GstPushSrc * src, GstBuffer * outbuf)
{
GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src);
GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
GstFlowReturn ret = GST_FLOW_OK;
GstMapInfo info;
SRTSOCKET ready[2];
gint recv_len;
if (srt_epoll_wait (priv->poll_id, 0, 0, ready, &(int) {
2}, priv->poll_timeout, 0, 0, 0, 0) == -1) {
/* Assuming that timeout error is normal */
if (srt_getlasterror (NULL) != SRT_ETIMEOUT) {
GST_ELEMENT_ERROR (src, RESOURCE, READ,
(NULL), ("srt_epoll_wait error: %s", srt_getlasterror_str ()));
ret = GST_FLOW_ERROR;
}
srt_clearlasterror ();
goto out;
}
if (!gst_buffer_map (outbuf, &info, GST_MAP_WRITE)) {
GST_ELEMENT_ERROR (src, RESOURCE, READ,
("Could not map the buffer for writing "), (NULL));
ret = GST_FLOW_ERROR;
goto out;
}
recv_len = srt_recvmsg (priv->sock, (char *) info.data,
gst_buffer_get_size (outbuf));
gst_buffer_unmap (outbuf, &info);
if (recv_len == SRT_ERROR) {
GST_ELEMENT_ERROR (src, RESOURCE, READ,
(NULL), ("srt_recvmsg error: %s", srt_getlasterror_str ()));
ret = GST_FLOW_ERROR;
goto out;
} else if (recv_len == 0) {
ret = GST_FLOW_EOS;
goto out;
}
GST_BUFFER_PTS (outbuf) =
gst_clock_get_time (GST_ELEMENT_CLOCK (src)) -
GST_ELEMENT_CAST (src)->base_time;
gst_buffer_resize (outbuf, 0, recv_len);
GST_LOG_OBJECT (src,
"filled buffer from _get of size %" G_GSIZE_FORMAT ", ts %"
GST_TIME_FORMAT ", dur %" GST_TIME_FORMAT
", offset %" G_GINT64_FORMAT ", offset_end %" G_GINT64_FORMAT,
gst_buffer_get_size (outbuf),
GST_TIME_ARGS (GST_BUFFER_TIMESTAMP (outbuf)),
GST_TIME_ARGS (GST_BUFFER_DURATION (outbuf)),
GST_BUFFER_OFFSET (outbuf), GST_BUFFER_OFFSET_END (outbuf));
out:
return ret;
}
static gboolean
gst_srt_client_src_start (GstBaseSrc * src)
{
GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src);
GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
GstSRTBaseSrc *base = GST_SRT_BASE_SRC (src);
GstUri *uri = gst_uri_ref (base->uri);
GSocketAddress *socket_address = NULL;
priv->sock = gst_srt_client_connect_full (GST_ELEMENT (src), FALSE,
gst_uri_get_host (uri), gst_uri_get_port (uri), priv->rendez_vous,
priv->bind_address, priv->bind_port, base->latency,
&socket_address, &priv->poll_id, base->passphrase, base->key_length);
g_clear_object (&socket_address);
g_clear_pointer (&uri, gst_uri_unref);
return (priv->sock != SRT_INVALID_SOCK);
}
static gboolean
gst_srt_client_src_stop (GstBaseSrc * src)
{
GstSRTClientSrc *self = GST_SRT_CLIENT_SRC (src);
GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
if (priv->poll_id != SRT_ERROR) {
if (priv->sock != SRT_INVALID_SOCK)
srt_epoll_remove_usock (priv->poll_id, priv->sock);
srt_epoll_release (priv->poll_id);
}
priv->poll_id = SRT_ERROR;
GST_DEBUG_OBJECT (self, "closing SRT connection");
if (priv->sock != SRT_INVALID_SOCK)
srt_close (priv->sock);
priv->sock = SRT_INVALID_SOCK;
return TRUE;
}
static void
gst_srt_client_src_class_init (GstSRTClientSrcClass * klass)
{
GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
GstBaseSrcClass *gstbasesrc_class = GST_BASE_SRC_CLASS (klass);
GstPushSrcClass *gstpushsrc_class = GST_PUSH_SRC_CLASS (klass);
gobject_class->set_property = gst_srt_client_src_set_property;
gobject_class->get_property = gst_srt_client_src_get_property;
gobject_class->finalize = gst_srt_client_src_finalize;
/**
* GstSRTClientSrc:poll-timeout:
*
* The timeout(ms) value when polling SRT socket.
*/
properties[PROP_POLL_TIMEOUT] =
g_param_spec_int ("poll-timeout", "Poll timeout",
"Return poll wait after timeout miliseconds (-1 = infinite)", -1,
G_MAXINT32, SRT_DEFAULT_POLL_TIMEOUT,
G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS);
properties[PROP_BIND_ADDRESS] =
g_param_spec_string ("bind-address", "Bind Address",
"Address to bind socket to (required for rendez-vous mode) ", NULL,
G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS);
properties[PROP_BIND_PORT] =
g_param_spec_int ("bind-port", "Bind Port",
"Port to bind socket to (Ignored in rendez-vous mode)", 0,
G_MAXUINT16, 0,
G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS);
properties[PROP_RENDEZ_VOUS] =
g_param_spec_boolean ("rendez-vous", "Rendez Vous",
"Work in Rendez-Vous mode instead of client/caller mode", FALSE,
G_PARAM_READWRITE | GST_PARAM_MUTABLE_READY | G_PARAM_STATIC_STRINGS);
g_object_class_install_properties (gobject_class, PROP_LAST, properties);
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gst_element_class_set_metadata (gstelement_class,
"SRT client source", "Source/Network",
"Receive data over the network via SRT",
"Justin Kim <justin.kim@collabora.com>");
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_srt_client_src_start);
gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_srt_client_src_stop);
gstpushsrc_class->fill = GST_DEBUG_FUNCPTR (gst_srt_client_src_fill);
}
static void
gst_srt_client_src_init (GstSRTClientSrc * self)
{
GstSRTClientSrcPrivate *priv = GST_SRT_CLIENT_SRC_GET_PRIVATE (self);
priv->sock = SRT_INVALID_SOCK;
priv->poll_id = SRT_ERROR;
priv->poll_timeout = SRT_DEFAULT_POLL_TIMEOUT;
priv->rendez_vous = FALSE;
priv->bind_address = NULL;
priv->bind_port = 0;
}