/* 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 "gstsrtelements.h" #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, SIG_CALLER_REJECTED, SIG_CALLER_CONNECTING, LAST_SIGNAL }; enum { PROP_KEEP_LISTENING = 128 }; 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); static gboolean src_default_caller_connecting (GstSRTSrc * self, GSocketAddress * addr, const gchar * username, gpointer data); static gboolean src_authentication_accumulator (GSignalInvocationHint * ihint, GValue * return_accu, const GValue * handler_return, gpointer data); #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")); GST_ELEMENT_REGISTER_DEFINE_WITH_CODE (srtsrc, "srtsrc", GST_RANK_PRIMARY, GST_TYPE_SRT_SRC, srt_element_init (plugin)); static gboolean src_default_caller_connecting (GstSRTSrc * self, GSocketAddress * addr, const gchar * stream_id, gpointer data) { /* Accept all connections. */ return TRUE; } static gboolean src_authentication_accumulator (GSignalInvocationHint * ihint, GValue * return_accu, const GValue * handler_return, gpointer data) { gboolean ret = g_value_get_boolean (handler_return); /* Handlers return TRUE on authentication success and we want to stop on * the first failure. */ g_value_set_boolean (return_accu, ret); return ret; } static gboolean gst_srt_src_start (GstBaseSrc * bsrc) { GstSRTSrc *self = GST_SRT_SRC (bsrc); GError *error = NULL; gboolean ret = FALSE; ret = gst_srt_object_open (self->srtobject, &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); } /* Reset expected pktseq */ self->next_pktseq = 0; 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; GstClock *clock; GstClockTime base_time; GstClockTime capture_time; GstClockTimeDiff delay; int64_t srt_time; SRT_MSGCTRL mctrl; retry: if (g_cancellable_is_cancelled (self->srtobject->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; } /* Get clock and values */ clock = gst_element_get_clock (GST_ELEMENT (src)); if (!clock) { GST_DEBUG_OBJECT (src, "Clock missing, flushing"); return GST_FLOW_FLUSHING; } base_time = gst_element_get_base_time (GST_ELEMENT (src)); recv_len = gst_srt_object_read (self->srtobject, info.data, gst_buffer_get_size (outbuf), &err, &mctrl); /* Capture clock values ASAP */ capture_time = gst_clock_get_time (clock); #if SRT_VERSION_VALUE >= 0x10402 /* Use SRT clock value if available (SRT > 1.4.2) */ srt_time = srt_time_now (); #else /* Else use the unix epoch monotonic clock */ srt_time = g_get_real_time (); #endif gst_object_unref (clock); gst_buffer_unmap (outbuf, &info); GST_LOG_OBJECT (src, "recv_len:%" G_GSIZE_FORMAT " pktseq:%d msgno:%d srctime:%" G_GINT64_FORMAT, recv_len, mctrl.pktseq, mctrl.msgno, mctrl.srctime); if (g_cancellable_is_cancelled (self->srtobject->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) { gst_srt_src_stop (GST_BASE_SRC (self)); if (self->keep_listening && gst_srt_src_start (GST_BASE_SRC (self))) { /* FIXME: Should send GAP event(s) downstream */ gst_element_post_message (GST_ELEMENT_CAST (self), gst_message_new_element (GST_OBJECT_CAST (self), gst_structure_new_empty ("connection-removed"))); goto retry; } else { ret = GST_FLOW_EOS; goto out; } } /* Detect discontinuities */ if (mctrl.pktseq != self->next_pktseq) { GST_WARNING_OBJECT (src, "discont detected %d (expected: %d)", mctrl.pktseq, self->next_pktseq); GST_BUFFER_FLAG_SET (outbuf, GST_BUFFER_FLAG_DISCONT); } /* pktseq is a 31bit field */ self->next_pktseq = (mctrl.pktseq + 1) % G_MAXINT32; /* 0 means we do not have a srctime */ if (mctrl.srctime != 0) delay = (srt_time - mctrl.srctime) * GST_USECOND; else delay = 0; GST_LOG_OBJECT (src, "delay: %" GST_STIME_FORMAT, GST_STIME_ARGS (delay)); if (delay < 0) { GST_WARNING_OBJECT (src, "Calculated SRT delay %" GST_STIME_FORMAT " is negative, clamping to 0", GST_STIME_ARGS (delay)); delay = 0; } /* Subtract the base_time (since the pipeline started) ... */ if (capture_time > base_time) capture_time -= base_time; else capture_time = 0; /* And adjust by the delay */ if (capture_time > delay) capture_time -= delay; else capture_time = 0; GST_BUFFER_TIMESTAMP (outbuf) = capture_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 void gst_srt_src_init (GstSRTSrc * self) { self->srtobject = gst_srt_object_new (GST_ELEMENT (self)); gst_base_src_set_format (GST_BASE_SRC (self), GST_FORMAT_TIME); gst_base_src_set_live (GST_BASE_SRC (self), TRUE); /* We do the timing ourselves */ gst_base_src_set_do_timestamp (GST_BASE_SRC (self), FALSE); 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); 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_unlock (self->srtobject); return TRUE; } static gboolean gst_srt_src_unlock_stop (GstBaseSrc * bsrc) { GstSRTSrc *self = GST_SRT_SRC (bsrc); gst_srt_object_unlock_stop (self->srtobject); 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)) { switch (prop_id) { case PROP_KEEP_LISTENING: self->keep_listening = g_value_get_boolean (value); break; default: 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)) { switch (prop_id) { case PROP_KEEP_LISTENING: g_value_set_boolean (value, self->keep_listening); break; default: G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec); } } } static gboolean gst_srt_src_query (GstBaseSrc * basesrc, GstQuery * query) { GstSRTSrc *self = GST_SRT_SRC (basesrc); if (GST_QUERY_TYPE (query) == GST_QUERY_LATENCY) { gint latency; if (!gst_structure_get_int (self->srtobject->parameters, "latency", &latency)) latency = GST_SRT_DEFAULT_LATENCY; gst_query_set_latency (query, TRUE, latency * GST_MSECOND, latency * GST_MSECOND); return TRUE; } else { return GST_BASE_SRC_CLASS (parent_class)->query (basesrc, query); } } 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; klass->caller_connecting = src_default_caller_connecting; /** * GstSRTSrc::caller-added: * @gstsrtsrc: the srtsrc element that emitted this signal * @unused: always zero (for ABI compatibility with previous versions) * @addr: the #GSocketAddress of the new caller * * A new caller has connected to srtsrc. */ 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: * @gstsrtsrc: the srtsrc element that emitted this signal * @unused: always zero (for ABI compatibility with previous versions) * @addr: the #GSocketAddress of the caller * * The given caller has disconnected. */ 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); /** * GstSRTSrc::caller-rejected: * @gstsrtsrc: the srtsrc element that emitted this signal * @addr: the #GSocketAddress that describes the client socket * @stream_id: the stream Id to which the caller wants to connect * * A caller's connection to srtsrc in listener mode has been rejected. * * Since: 1.20 * */ signals[SIG_CALLER_REJECTED] = g_signal_new ("caller-rejected", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass, caller_rejected), NULL, NULL, NULL, G_TYPE_NONE, 2, G_TYPE_SOCKET_ADDRESS, G_TYPE_STRING); /** * GstSRTSrc::caller-connecting: * @gstsrtsrc: the srtsrc element that emitted this signal * @addr: the #GSocketAddress that describes the client socket * @stream_id: the stream Id to which the caller wants to connect * * Whether to accept or reject a caller's connection to srtsrc in listener mode. * The Caller's connection is rejected if the callback returns FALSE, else * the connection is accepeted. * * Since: 1.20 * */ signals[SIG_CALLER_CONNECTING] = g_signal_new ("caller-connecting", G_TYPE_FROM_CLASS (klass), G_SIGNAL_RUN_LAST, G_STRUCT_OFFSET (GstSRTSrcClass, caller_connecting), src_authentication_accumulator, NULL, NULL, G_TYPE_BOOLEAN, 2, G_TYPE_SOCKET_ADDRESS, G_TYPE_STRING); gst_srt_object_install_properties_helper (gobject_class); /** * GstSRTSrc:keep-listening: * * If FALSE, the element will return GST_FLOW_EOS when the remote client disconnects. * If TRUE, the element will keep waiting for the client to reconnect. An element * message named 'connection-removed' will be sent on disconnection. * * Since: 1.22 * */ g_object_class_install_property (gobject_class, PROP_KEEP_LISTENING, g_param_spec_boolean ("keep-listening", "Keep listening", "Toggle keep-listening for connection reuse", FALSE, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS)); 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); gstbasesrc_class->query = GST_DEBUG_FUNCPTR (gst_srt_src_query); 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); gboolean ret; GST_OBJECT_LOCK (self); ret = gst_srt_object_set_uri (self->srtobject, uri, error); GST_OBJECT_UNLOCK (self); return ret; } 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; }