gstreamer/ext/srt/gstsrtsrc.c
Justin Kim 0a350c610d srt: Integrate server and client element into one
We have srt{client,server}{src,sink} elements in accordance to the
norm of the connection oriented protocols. However, SRT connection
mode can be changed by uri parameters so it requires an integrated
element to handle the parameters.

fix: #740
2019-01-09 19:44:02 +00:00

360 lines
10 KiB
C

/* GStreamer
* Copyright (C) 2018, Collabora Ltd.
* Copyright (C) 2018, SK Telecom, Co., Ltd.
* Author: Jeongseok Kim <jeongseok.kim@sk.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-srtsrc
* @title: srtsrc
*
* srtsrc is a network source that reads <ulink url="http://www.srtalliance.org/">SRT</ulink>
* packets from the network.
*
* <refsect2>
* <title>Examples</title>
* |[
* gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:7001" ! fakesink
* ]| This pipeline shows how to connect SRT server by setting #GstSRTSrc:uri property.
*
* |[
* gst-launch-1.0 -v srtsrc uri="srt://127.0.0.1:7001?mode=listener" ! fakesink
* ]| This pipeline shows how to wait SRT connection by setting #GstSRTSrc:uri property.
*
* |[
* gst-launch-1.0 -v srtclientsrc uri="srt://192.168.1.10:7001?mode=rendez-vous" ! fakesink
* ]| This pipeline shows how to connect SRT server by setting #GstSRTSrc:uri property and using the rendez-vous mode.
* </refsect2>
*
*/
#ifdef HAVE_CONFIG_H
#include <config.h>
#endif
#include "gstsrtsrc.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_src
GST_DEBUG_CATEGORY (GST_CAT_DEFAULT);
enum
{
SIG_CALLER_ADDED,
SIG_CALLER_REMOVED,
LAST_SIGNAL
};
static guint signals[LAST_SIGNAL] = { 0 };
static void gst_srt_src_uri_handler_init (gpointer g_iface,
gpointer iface_data);
static gchar *gst_srt_src_uri_get_uri (GstURIHandler * handler);
static gboolean gst_srt_src_uri_set_uri (GstURIHandler * handler,
const gchar * uri, GError ** error);
#define gst_srt_src_parent_class parent_class
G_DEFINE_TYPE_WITH_CODE (GstSRTSrc, gst_srt_src,
GST_TYPE_PUSH_SRC,
G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_srt_src_uri_handler_init)
GST_DEBUG_CATEGORY_INIT (GST_CAT_DEFAULT, "srtsrc", 0, "SRT Source"));
static void
gst_srt_src_caller_added_cb (int sock, GSocketAddress * addr,
GstSRTObject * srtobject)
{
g_signal_emit (srtobject->element, signals[SIG_CALLER_ADDED], 0, sock, addr);
}
static void
gst_srt_src_caller_removed_cb (int sock, GSocketAddress * addr,
GstSRTObject * srtobject)
{
g_signal_emit (srtobject->element, signals[SIG_CALLER_REMOVED], 0, sock,
addr);
}
static gboolean
gst_srt_src_start (GstBaseSrc * bsrc)
{
GstSRTSrc *self = GST_SRT_SRC (bsrc);
GError *error = NULL;
gboolean ret = FALSE;
GstSRTConnectionMode connection_mode = GST_SRT_CONNECTION_MODE_NONE;
gst_structure_get_enum (self->srtobject->parameters, "mode",
GST_TYPE_SRT_CONNECTION_MODE, (gint *) & connection_mode);
if (connection_mode == GST_SRT_CONNECTION_MODE_LISTENER) {
ret =
gst_srt_object_open_full (self->srtobject, gst_srt_src_caller_added_cb,
gst_srt_src_caller_removed_cb, self->cancellable, &error);
} else {
ret = gst_srt_object_open (self->srtobject, self->cancellable, &error);
}
if (!ret) {
GST_WARNING_OBJECT (self, "Failed to open SRT: %s", error->message);
g_clear_error (&error);
}
return ret;
}
static gboolean
gst_srt_src_stop (GstBaseSrc * bsrc)
{
GstSRTSrc *self = GST_SRT_SRC (bsrc);
gst_srt_object_close (self->srtobject);
return TRUE;
}
static GstFlowReturn
gst_srt_src_fill (GstPushSrc * src, GstBuffer * outbuf)
{
GstSRTSrc *self = GST_SRT_SRC (src);
GstFlowReturn ret = GST_FLOW_OK;
GstMapInfo info;
GError *err = NULL;
gssize recv_len;
if (g_cancellable_is_cancelled (self->cancellable)) {
ret = GST_FLOW_FLUSHING;
}
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 = gst_srt_object_read (self->srtobject, info.data,
gst_buffer_get_size (outbuf), self->cancellable, &err);
gst_buffer_unmap (outbuf, &info);
if (g_cancellable_is_cancelled (self->cancellable)) {
ret = GST_FLOW_FLUSHING;
goto out;
}
if (recv_len < 0) {
GST_ELEMENT_ERROR (src, RESOURCE, READ, (NULL), ("%s", err->message));
ret = GST_FLOW_ERROR;
g_clear_error (&err);
goto out;
} else if (recv_len == 0) {
ret = GST_FLOW_EOS;
goto out;
}
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 void
gst_srt_src_init (GstSRTSrc * self)
{
self->srtobject = gst_srt_object_new (GST_ELEMENT (self));
self->cancellable = g_cancellable_new ();
gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME);
gst_base_src_set_live (GST_BASE_SRC (self), TRUE);
gst_base_src_set_do_timestamp (GST_BASE_SRC (self), TRUE);
gst_srt_object_set_uri (self->srtobject, GST_SRT_DEFAULT_URI, NULL);
}
static void
gst_srt_src_finalize (GObject * object)
{
GstSRTSrc *self = GST_SRT_SRC (object);
g_clear_object (&self->cancellable);
gst_srt_object_destroy (self->srtobject);
G_OBJECT_CLASS (parent_class)->finalize (object);
}
static gboolean
gst_srt_src_unlock (GstBaseSrc * bsrc)
{
GstSRTSrc *self = GST_SRT_SRC (bsrc);
g_cancellable_cancel (self->cancellable);
gst_srt_object_wakeup (self->srtobject);
return TRUE;
}
static gboolean
gst_srt_src_unlock_stop (GstBaseSrc * bsrc)
{
GstSRTSrc *self = GST_SRT_SRC (bsrc);
g_cancellable_reset (self->cancellable);
return TRUE;
}
static void
gst_srt_src_set_property (GObject * object,
guint prop_id, const GValue * value, GParamSpec * pspec)
{
GstSRTSrc *self = GST_SRT_SRC (object);
if (!gst_srt_object_set_property_helper (self->srtobject, prop_id, value,
pspec)) {
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
}
static void
gst_srt_src_get_property (GObject * object,
guint prop_id, GValue * value, GParamSpec * pspec)
{
GstSRTSrc *self = GST_SRT_SRC (object);
if (!gst_srt_object_get_property_helper (self->srtobject, prop_id, value,
pspec)) {
G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
}
}
static void
gst_srt_src_class_init (GstSRTSrcClass * 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_src_set_property;
gobject_class->get_property = gst_srt_src_get_property;
gobject_class->finalize = gst_srt_src_finalize;
/**
* GstSRTSrc::caller-added:
* @gstsrtsink: the srtsink element that emitted this signal
* @sock: the client socket descriptor that was added to srtsink
* @addr: the #GSocketAddress that describes the @sock
*
* The given socket descriptor was added to srtsink.
*/
signals[SIG_CALLER_ADDED] =
g_signal_new ("caller-added", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass, caller_added),
NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE,
2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
/**
* GstSRTSrc::caller-removed:
* @gstsrtsink: the srtsink element that emitted this signal
* @sock: the client socket descriptor that was added to srtsink
* @addr: the #GSocketAddress that describes the @sock
*
* The given socket descriptor was removed from srtsink.
*/
signals[SIG_CALLER_REMOVED] =
g_signal_new ("caller-removed", G_TYPE_FROM_CLASS (klass),
G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass,
caller_added), NULL, NULL, g_cclosure_marshal_generic, G_TYPE_NONE,
2, G_TYPE_INT, G_TYPE_SOCKET_ADDRESS);
gst_srt_object_install_properties_helper (gobject_class);
gst_element_class_add_static_pad_template (gstelement_class, &src_template);
gst_element_class_set_metadata (gstelement_class,
"SRT source", "Source/Network",
"Receive data over the network via SRT",
"Justin Kim <justin.joy.9to5@gmail.com>");
gstbasesrc_class->start = GST_DEBUG_FUNCPTR (gst_srt_src_start);
gstbasesrc_class->stop = GST_DEBUG_FUNCPTR (gst_srt_src_stop);
gstbasesrc_class->unlock = GST_DEBUG_FUNCPTR (gst_srt_src_unlock);
gstbasesrc_class->unlock_stop = GST_DEBUG_FUNCPTR (gst_srt_src_unlock_stop);
gstpushsrc_class->fill = GST_DEBUG_FUNCPTR (gst_srt_src_fill);
}
static GstURIType
gst_srt_src_uri_get_type (GType type)
{
return GST_URI_SRC;
}
static const gchar *const *
gst_srt_src_uri_get_protocols (GType type)
{
static const gchar *protocols[] = { GST_SRT_DEFAULT_URI_SCHEME, NULL };
return protocols;
}
static gchar *
gst_srt_src_uri_get_uri (GstURIHandler * handler)
{
gchar *uri_str;
GstSRTSrc *self = GST_SRT_SRC (handler);
GST_OBJECT_LOCK (self);
uri_str = gst_uri_to_string (self->srtobject->uri);
GST_OBJECT_UNLOCK (self);
return uri_str;
}
static gboolean
gst_srt_src_uri_set_uri (GstURIHandler * handler,
const gchar * uri, GError ** error)
{
GstSRTSrc *self = GST_SRT_SRC (handler);
return gst_srt_object_set_uri (self->srtobject, uri, error);
}
static void
gst_srt_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
{
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
iface->get_type = gst_srt_src_uri_get_type;
iface->get_protocols = gst_srt_src_uri_get_protocols;
iface->get_uri = gst_srt_src_uri_get_uri;
iface->set_uri = gst_srt_src_uri_set_uri;
}