/* GStreamer * Copyright (C) 2018, Collabora Ltd. * Copyright (C) 2018, SK Telecom, Co., Ltd. * Author: Jeongseok Kim * * 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 [SRT](http://www.srtalliance.org/) * packets from the network. * * ## Examples * |[ * 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://: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. * */ #ifdef HAVE_CONFIG_H #include #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) { /* ensure error is posted since state change will fail */ GST_ELEMENT_ERROR (self, RESOURCE, OPEN_READ, (NULL), ("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); gst_srt_object_wakeup (self->srtobject, self->cancellable); 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, NULL, 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, NULL, 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 "); 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; }