/* GStreamer SRT plugin based on libsrt * Copyright (C) 2017, Collabora Ltd. * Author:Justin 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-srtclientsrc * @title: srtclientsrc * * srtclientsrc is a network source that reads SRT * packets from the network. Although SRT is a protocol based on UDP, srtclientsrc works like * a client socket of connection-oriented protocol. * * * Examples * |[ * 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. * * */ #ifdef HAVE_CONFIG_H #include "config.h" #endif #include "gstsrtclientsrc.h" #include #include #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 "); 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; }