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https://gitlab.freedesktop.org/gstreamer/gstreamer.git
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774 lines
23 KiB
C
774 lines
23 KiB
C
/* GStreamer
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* Copyright (C) <2018> Marc Leeman <marc.leeman@gmail.com>
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*
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* This library is free software; you can redistribute it and/or
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* modify it under the terms of the GNU Library General Public
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* License as published by the Free Software Foundation; either
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* version 2 of the License, or (at your option) any later version.
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*
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* This library is distributed in the hope that it will be useful,
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* but WITHOUT ANY WARRANTY; without even the implied warranty of
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* MERCHANTABILITY or FITNESS FOR A PARTICULAR PURPOSE. See the GNU
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* Library General Public License for more details.
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*
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* You should have received a copy of the GNU Library General Public
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* License along with this library; if not, write to the
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* Free Software Foundation, Inc., 51 Franklin St, Fifth Floor,
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* Boston, MA 02110-1301, USA.
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*/
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/**
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* SECTION: gstrtpsrc
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* @title: GstRtpSrc
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* @short description: element with Uri interface to get RTP data from
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* the network.
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*
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* RTP (RFC 3550) is a protocol to stream media over the network while
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* retaining the timing information and providing enough information to
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* reconstruct the correct timing domain by the receiver.
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*
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* The RTP data port should be even, while the RTCP port should be
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* odd. The URI that is entered defines the data port, the RTCP port will
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* be allocated to the next port.
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*
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* This element hooks up the correct sockets to support both RTP as the
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* accompanying RTCP layer.
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*
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* This Bin handles taking in of data from the network and provides the
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* RTP payloaded data.
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*
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* This element also implements the URI scheme `rtp://` allowing to render
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* RTP streams in GStreamer based media players. The RTP URI handler also
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* allows setting properties through the URI query.
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*/
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#ifdef HAVE_CONFIG_H
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#include <config.h>
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#endif
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#include <gst/net/net.h>
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#include <gst/rtp/gstrtppayloads.h>
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#include "gstrtpsrc.h"
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#include "gstrtp-utils.h"
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GST_DEBUG_CATEGORY_STATIC (gst_rtp_src_debug);
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#define GST_CAT_DEFAULT gst_rtp_src_debug
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#define DEFAULT_PROP_TTL 64
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#define DEFAULT_PROP_TTL_MC 1
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#define DEFAULT_PROP_ENCODING_NAME NULL
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#define DEFAULT_PROP_LATENCY 200
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#define DEFAULT_PROP_ADDRESS "0.0.0.0"
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#define DEFAULT_PROP_PORT 5004
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#define DEFAULT_PROP_URI "rtp://"DEFAULT_PROP_ADDRESS":"G_STRINGIFY(DEFAULT_PROP_PORT)
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enum
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{
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PROP_0,
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PROP_URI,
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PROP_ADDRESS,
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PROP_PORT,
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PROP_TTL,
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PROP_TTL_MC,
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PROP_ENCODING_NAME,
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PROP_LATENCY,
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PROP_LAST
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};
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static void gst_rtp_src_uri_handler_init (gpointer g_iface,
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gpointer iface_data);
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#define gst_rtp_src_parent_class parent_class
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G_DEFINE_TYPE_WITH_CODE (GstRtpSrc, gst_rtp_src, GST_TYPE_BIN,
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G_IMPLEMENT_INTERFACE (GST_TYPE_URI_HANDLER, gst_rtp_src_uri_handler_init);
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GST_DEBUG_CATEGORY_INIT (gst_rtp_src_debug, "rtpsrc", 0, "RTP Source"));
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#define GST_RTP_SRC_GET_LOCK(obj) (&((GstRtpSrc*)(obj))->lock)
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#define GST_RTP_SRC_LOCK(obj) (g_mutex_lock (GST_RTP_SRC_GET_LOCK(obj)))
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#define GST_RTP_SRC_UNLOCK(obj) (g_mutex_unlock (GST_RTP_SRC_GET_LOCK(obj)))
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static GstStaticPadTemplate src_template = GST_STATIC_PAD_TEMPLATE ("src_%u",
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GST_PAD_SRC,
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GST_PAD_SOMETIMES,
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GST_STATIC_CAPS ("application/x-rtp"));
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static GstStateChangeReturn
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gst_rtp_src_change_state (GstElement * element, GstStateChange transition);
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/**
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* gst_rtp_src_rtpbin_request_pt_map_cb:
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* @self: The current #GstRtpSrc object
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*
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* #GstRtpBin callback to map a pt on RTP caps.
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*
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* Returns: (transfer none): the guess on the RTP caps based on the PT
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* and caps.
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*/
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static GstCaps *
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gst_rtp_src_rtpbin_request_pt_map_cb (GstElement * rtpbin, guint session_id,
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guint pt, gpointer data)
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{
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GstRtpSrc *self = GST_RTP_SRC (data);
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const GstRTPPayloadInfo *p = NULL;
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GST_DEBUG_OBJECT (self,
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"Requesting caps for session-id 0x%x and pt %u.", session_id, pt);
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/* the encoding-name has more relevant information */
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if (self->encoding_name != NULL) {
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/* Unfortunately, the media needs to be passed in the function. Since
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* it is not known, try for video if video not found. */
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p = gst_rtp_payload_info_for_name ("video", self->encoding_name);
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if (p == NULL)
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p = gst_rtp_payload_info_for_name ("audio", self->encoding_name);
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}
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/* If info has been found before based on the encoding-name, go with
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* it. If not, try to look it up on with a static one. Needs to be guarded
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* because some encoders do not use dynamic values for H.264 */
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if (p == NULL) {
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/* Static payload types, this is a simple lookup */
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if (!GST_RTP_PAYLOAD_IS_DYNAMIC (pt)) {
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p = gst_rtp_payload_info_for_pt (pt);
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}
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}
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if (p != NULL) {
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GstCaps *ret = gst_caps_new_simple ("application/x-rtp",
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"encoding-name", G_TYPE_STRING, p->encoding_name,
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"clock-rate", G_TYPE_INT, p->clock_rate,
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"media", G_TYPE_STRING, p->media, NULL);
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GST_DEBUG_OBJECT (self, "Decided on caps %" GST_PTR_FORMAT, ret);
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return ret;
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}
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GST_DEBUG_OBJECT (self, "Could not determine caps based on pt and"
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" the encoding-name was not set.");
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return NULL;
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}
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static void
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gst_rtp_src_set_property (GObject * object, guint prop_id,
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const GValue * value, GParamSpec * pspec)
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{
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GstRtpSrc *self = GST_RTP_SRC (object);
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GstCaps *caps;
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switch (prop_id) {
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case PROP_URI:{
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GstUri *uri = NULL;
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GST_RTP_SRC_LOCK (object);
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uri = gst_uri_from_string (g_value_get_string (value));
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if (uri == NULL)
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break;
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if (self->uri)
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gst_uri_unref (self->uri);
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self->uri = uri;
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/* Recursive set to self, do not use the same lock in all property
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* setters. */
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g_object_set (self, "address", gst_uri_get_host (self->uri), NULL);
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g_object_set (self, "port", gst_uri_get_port (self->uri), NULL);
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gst_rtp_utils_set_properties_from_uri_query (G_OBJECT (self), self->uri);
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GST_RTP_SRC_UNLOCK (object);
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break;
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}
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case PROP_ADDRESS:{
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GInetAddress *addr;
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gst_uri_set_host (self->uri, g_value_get_string (value));
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g_object_set_property (G_OBJECT (self->rtp_src), "address", value);
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addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
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if (g_inet_address_get_is_multicast (addr)) {
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g_object_set (self->rtcp_src, "address", gst_uri_get_host (self->uri),
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NULL);
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}
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g_object_unref (addr);
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break;
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}
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case PROP_PORT:{
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guint port = g_value_get_uint (value);
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/* According to RFC 3550, 11, RTCP receiver port should be even
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* number and RTCP port should be the RTP port + 1 */
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if (port & 0x1)
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GST_WARNING_OBJECT (self,
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"Port %u is odd, this is not standard (see RFC 3550).", port);
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gst_uri_set_port (self->uri, port);
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g_object_set (self->rtp_src, "port", port, NULL);
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g_object_set (self->rtcp_src, "port", port + 1, NULL);
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break;
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}
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case PROP_TTL:
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self->ttl = g_value_get_int (value);
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break;
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case PROP_TTL_MC:
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self->ttl_mc = g_value_get_int (value);
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break;
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case PROP_ENCODING_NAME:
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g_free (self->encoding_name);
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self->encoding_name = g_value_dup_string (value);
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if (self->rtp_src) {
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caps = gst_rtp_src_rtpbin_request_pt_map_cb (NULL, 0, 96, self);
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g_object_set (G_OBJECT (self->rtp_src), "caps", caps, NULL);
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gst_caps_unref (caps);
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}
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break;
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case PROP_LATENCY:
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g_object_set (self->rtpbin, "latency", g_value_get_uint (value), NULL);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_src_get_property (GObject * object, guint prop_id,
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GValue * value, GParamSpec * pspec)
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{
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GstRtpSrc *self = GST_RTP_SRC (object);
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switch (prop_id) {
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case PROP_URI:
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GST_RTP_SRC_LOCK (object);
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if (self->uri)
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g_value_take_string (value, gst_uri_to_string (self->uri));
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else
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g_value_set_string (value, NULL);
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GST_RTP_SRC_UNLOCK (object);
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break;
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case PROP_ADDRESS:
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g_value_set_string (value, gst_uri_get_host (self->uri));
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break;
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case PROP_PORT:
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g_value_set_uint (value, gst_uri_get_port (self->uri));
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break;
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case PROP_TTL:
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g_value_set_int (value, self->ttl);
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break;
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case PROP_TTL_MC:
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g_value_set_int (value, self->ttl_mc);
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break;
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case PROP_ENCODING_NAME:
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g_value_set_string (value, self->encoding_name);
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break;
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case PROP_LATENCY:
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g_object_get_property (G_OBJECT (self->rtpbin), "latency", value);
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break;
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default:
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G_OBJECT_WARN_INVALID_PROPERTY_ID (object, prop_id, pspec);
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break;
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}
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}
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static void
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gst_rtp_src_finalize (GObject * gobject)
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{
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GstRtpSrc *self = GST_RTP_SRC (gobject);
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if (self->uri)
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gst_uri_unref (self->uri);
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g_free (self->encoding_name);
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g_mutex_clear (&self->lock);
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G_OBJECT_CLASS (parent_class)->finalize (gobject);
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}
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static void
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gst_rtp_src_class_init (GstRtpSrcClass * klass)
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{
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GObjectClass *gobject_class = G_OBJECT_CLASS (klass);
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GstElementClass *gstelement_class = GST_ELEMENT_CLASS (klass);
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gobject_class->set_property = gst_rtp_src_set_property;
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gobject_class->get_property = gst_rtp_src_get_property;
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gobject_class->finalize = gst_rtp_src_finalize;
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gstelement_class->change_state = gst_rtp_src_change_state;
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/**
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* GstRtpSrc:uri:
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*
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* uri to an RTP from. All GStreamer parameters can be
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* encoded in the URI, this URI format is RFC compliant.
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*/
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g_object_class_install_property (gobject_class, PROP_URI,
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g_param_spec_string ("uri", "URI",
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"URI in the form of rtp://host:port?query", DEFAULT_PROP_URI,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpSrc:address:
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*
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* Address to receive packets from (can be IPv4 or IPv6).
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*/
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g_object_class_install_property (gobject_class, PROP_ADDRESS,
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g_param_spec_string ("address", "Address",
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"Address to receive packets from (can be IPv4 or IPv6).",
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DEFAULT_PROP_ADDRESS, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpSrc:port:
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*
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* The port to listen to RTP packets, the RTCP port is this value
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* +1. This port must be an even number.
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*/
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g_object_class_install_property (gobject_class, PROP_PORT,
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g_param_spec_uint ("port", "Port", "The port to listen for RTP packets, "
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"the RTCP port is this value + 1. This port must be an even number.",
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2, 65534, DEFAULT_PROP_PORT,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS | G_PARAM_CONSTRUCT));
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/**
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* GstRtpSrc:ttl:
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*
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* Set the unicast TTL parameter. In RTP this of importance for RTCP.
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*/
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g_object_class_install_property (gobject_class, PROP_TTL,
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g_param_spec_int ("ttl", "Unicast TTL",
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"Used for setting the unicast TTL parameter",
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0, 255, DEFAULT_PROP_TTL,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpSrc:ttl-mc:
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*
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* Set the multicast TTL parameter. In RTP this of importance for RTCP.
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*/
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g_object_class_install_property (gobject_class, PROP_TTL_MC,
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g_param_spec_int ("ttl-mc", "Multicast TTL",
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"Used for setting the multicast TTL parameter", 0, 255,
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DEFAULT_PROP_TTL_MC, G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpSrc:encoding-name:
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*
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* Set the encoding name of the stream to use. This is a short-hand for
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* the full caps and maps typically to the encoding-name in the RTP caps.
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*/
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g_object_class_install_property (gobject_class, PROP_ENCODING_NAME,
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g_param_spec_string ("encoding-name", "Caps encoding name",
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"Encoding name use to determine caps parameters",
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DEFAULT_PROP_ENCODING_NAME,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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/**
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* GstRtpSrc:latency:
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*
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* Set the size of the latency buffer in the
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* GstRtpBin/GstRtpJitterBuffer to compensate for network jitter.
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*/
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g_object_class_install_property (gobject_class, PROP_LATENCY,
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g_param_spec_uint ("latency", "Buffer latency in ms",
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"Default amount of ms to buffer in the jitterbuffers", 0,
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G_MAXUINT, DEFAULT_PROP_LATENCY,
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G_PARAM_READWRITE | G_PARAM_STATIC_STRINGS));
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gst_element_class_add_pad_template (gstelement_class,
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gst_static_pad_template_get (&src_template));
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gst_element_class_set_static_metadata (gstelement_class,
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"RTP Source element",
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"Generic/Bin/Src",
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"Simple RTP src", "Marc Leeman <marc.leeman@gmail.com>");
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}
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static void
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gst_rtp_src_rtpbin_pad_added_cb (GstElement * element, GstPad * pad,
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gpointer data)
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{
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GstRtpSrc *self = GST_RTP_SRC (data);
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GstCaps *caps = gst_pad_query_caps (pad, NULL);
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GstPad *upad;
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gchar name[48];
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/* Expose RTP data pad only */
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GST_INFO_OBJECT (self,
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"Element %" GST_PTR_FORMAT " added pad %" GST_PTR_FORMAT "with caps %"
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GST_PTR_FORMAT ".", element, pad, caps);
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/* Sanity checks */
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if (GST_PAD_DIRECTION (pad) == GST_PAD_SINK) {
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/* Sink pad, do not expose */
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gst_caps_unref (caps);
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return;
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}
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if (G_LIKELY (caps)) {
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GstCaps *ref_caps = gst_caps_new_empty_simple ("application/x-rtcp");
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if (gst_caps_can_intersect (caps, ref_caps)) {
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/* SRC RTCP caps, do not expose */
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gst_caps_unref (ref_caps);
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gst_caps_unref (caps);
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return;
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}
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gst_caps_unref (ref_caps);
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} else {
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GST_ERROR_OBJECT (self, "Pad with no caps detected.");
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gst_caps_unref (caps);
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return;
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}
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gst_caps_unref (caps);
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GST_RTP_SRC_LOCK (self);
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g_snprintf (name, 48, "src_%u", GST_ELEMENT (self)->numpads);
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upad = gst_ghost_pad_new (name, pad);
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gst_pad_set_active (upad, TRUE);
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gst_element_add_pad (GST_ELEMENT (self), upad);
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GST_RTP_SRC_UNLOCK (self);
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}
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static void
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gst_rtp_src_rtpbin_pad_removed_cb (GstElement * element, GstPad * pad,
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gpointer data)
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{
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GstRtpSrc *self = GST_RTP_SRC (data);
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GST_INFO_OBJECT (self,
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"Element %" GST_PTR_FORMAT " removed pad %" GST_PTR_FORMAT ".", element,
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pad);
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}
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static void
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gst_rtp_src_rtpbin_on_ssrc_collision_cb (GstElement * rtpbin, guint session_id,
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guint ssrc, gpointer data)
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{
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GstRtpSrc *self = GST_RTP_SRC (data);
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GST_INFO_OBJECT (self,
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"Dectected an SSRC collision: session-id 0x%x, ssrc 0x%x.", session_id,
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ssrc);
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}
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static void
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gst_rtp_src_rtpbin_on_new_ssrc_cb (GstElement * rtpbin, guint session_id,
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guint ssrc, gpointer data)
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{
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GstRtpSrc *self = GST_RTP_SRC (data);
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GST_INFO_OBJECT (self, "Dectected a new SSRC: session-id 0x%x, ssrc 0x%x.",
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session_id, ssrc);
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
gst_rtp_src_on_recv_rtcp (GstPad * pad, GstPadProbeInfo * info,
|
|
gpointer user_data)
|
|
{
|
|
GstRtpSrc *self = GST_RTP_SRC (user_data);
|
|
GstBuffer *buffer;
|
|
GstNetAddressMeta *meta;
|
|
|
|
if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
|
|
GstBufferList *buffer_list = info->data;
|
|
buffer = gst_buffer_list_get (buffer_list, 0);
|
|
} else {
|
|
buffer = info->data;
|
|
}
|
|
|
|
meta = gst_buffer_get_net_address_meta (buffer);
|
|
|
|
GST_OBJECT_LOCK (self);
|
|
g_clear_object (&self->rtcp_send_addr);
|
|
self->rtcp_send_addr = g_object_ref (meta->addr);
|
|
GST_OBJECT_UNLOCK (self);
|
|
|
|
return GST_PAD_PROBE_OK;
|
|
}
|
|
|
|
static inline void
|
|
gst_rtp_src_attach_net_address_meta (GstRtpSrc * self, GstBuffer * buffer)
|
|
{
|
|
GST_OBJECT_LOCK (self);
|
|
if (self->rtcp_send_addr)
|
|
gst_buffer_add_net_address_meta (buffer, self->rtcp_send_addr);
|
|
GST_OBJECT_UNLOCK (self);
|
|
}
|
|
|
|
static GstPadProbeReturn
|
|
gst_rtp_src_on_send_rtcp (GstPad * pad, GstPadProbeInfo * info,
|
|
gpointer user_data)
|
|
{
|
|
GstRtpSrc *self = GST_RTP_SRC (user_data);
|
|
|
|
if (info->type == GST_PAD_PROBE_TYPE_BUFFER_LIST) {
|
|
GstBufferList *buffer_list = info->data;
|
|
GstBuffer *buffer;
|
|
gint i;
|
|
|
|
info->data = buffer_list = gst_buffer_list_make_writable (buffer_list);
|
|
for (i = 0; i < gst_buffer_list_length (buffer_list); i++) {
|
|
buffer = gst_buffer_list_get (buffer_list, i);
|
|
gst_rtp_src_attach_net_address_meta (self, buffer);
|
|
}
|
|
} else {
|
|
GstBuffer *buffer = info->data;
|
|
info->data = buffer = gst_buffer_make_writable (buffer);
|
|
gst_rtp_src_attach_net_address_meta (self, buffer);
|
|
}
|
|
|
|
return GST_PAD_PROBE_OK;
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_src_start (GstRtpSrc * self)
|
|
{
|
|
GstPad *pad;
|
|
GSocket *socket;
|
|
GInetAddress *addr;
|
|
GstCaps *caps;
|
|
|
|
/* Should not be NULL */
|
|
g_return_val_if_fail (self->uri != NULL, FALSE);
|
|
|
|
/* share the socket created by the source */
|
|
g_object_get (G_OBJECT (self->rtcp_src), "used-socket", &socket, NULL);
|
|
if (!G_IS_SOCKET (socket)) {
|
|
GST_WARNING_OBJECT (self, "Could not retrieve RTCP src socket.");
|
|
}
|
|
|
|
addr = g_inet_address_new_from_string (gst_uri_get_host (self->uri));
|
|
if (g_inet_address_get_is_multicast (addr)) {
|
|
/* mc-ttl is not supported by dynudpsink */
|
|
g_socket_set_multicast_ttl (socket, self->ttl_mc);
|
|
/* In multicast, send RTCP to the multicast group */
|
|
self->rtcp_send_addr =
|
|
g_inet_socket_address_new (addr, gst_uri_get_port (self->uri) + 1);
|
|
} else {
|
|
/* In unicast, send RTCP to the detected sender address */
|
|
g_socket_set_ttl (socket, self->ttl);
|
|
pad = gst_element_get_static_pad (self->rtcp_src, "src");
|
|
self->rtcp_recv_probe = gst_pad_add_probe (pad,
|
|
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
|
|
gst_rtp_src_on_recv_rtcp, self, NULL);
|
|
gst_object_unref (pad);
|
|
}
|
|
g_object_unref (addr);
|
|
|
|
/* no need to set address if unicast */
|
|
caps = gst_caps_new_empty_simple ("application/x-rtcp");
|
|
g_object_set (self->rtcp_src, "caps", caps, NULL);
|
|
gst_caps_unref (caps);
|
|
|
|
pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
|
|
self->rtcp_send_probe = gst_pad_add_probe (pad,
|
|
GST_PAD_PROBE_TYPE_BUFFER | GST_PAD_PROBE_TYPE_BUFFER_LIST,
|
|
gst_rtp_src_on_send_rtcp, self, NULL);
|
|
gst_object_unref (pad);
|
|
|
|
g_object_set (self->rtcp_sink, "socket", socket, "close-socket", FALSE, NULL);
|
|
g_object_unref (socket);
|
|
|
|
gst_element_set_locked_state (self->rtcp_sink, FALSE);
|
|
gst_element_sync_state_with_parent (self->rtcp_sink);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_src_stop (GstRtpSrc * self)
|
|
{
|
|
GstPad *pad;
|
|
|
|
if (self->rtcp_recv_probe) {
|
|
pad = gst_element_get_static_pad (self->rtcp_src, "src");
|
|
gst_pad_remove_probe (pad, self->rtcp_recv_probe);
|
|
self->rtcp_recv_probe = 0;
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
pad = gst_element_get_static_pad (self->rtcp_sink, "sink");
|
|
gst_pad_remove_probe (pad, self->rtcp_send_probe);
|
|
self->rtcp_send_probe = 0;
|
|
gst_object_unref (pad);
|
|
}
|
|
|
|
static GstStateChangeReturn
|
|
gst_rtp_src_change_state (GstElement * element, GstStateChange transition)
|
|
{
|
|
GstRtpSrc *self = GST_RTP_SRC (element);
|
|
GstStateChangeReturn ret = GST_STATE_CHANGE_SUCCESS;
|
|
|
|
GST_DEBUG_OBJECT (self, "Changing state: %s => %s",
|
|
gst_element_state_get_name (GST_STATE_TRANSITION_CURRENT (transition)),
|
|
gst_element_state_get_name (GST_STATE_TRANSITION_NEXT (transition)));
|
|
|
|
ret = GST_ELEMENT_CLASS (parent_class)->change_state (element, transition);
|
|
if (ret == GST_STATE_CHANGE_FAILURE)
|
|
return ret;
|
|
|
|
switch (transition) {
|
|
case GST_STATE_CHANGE_NULL_TO_READY:
|
|
if (gst_rtp_src_start (self) == FALSE)
|
|
return GST_STATE_CHANGE_FAILURE;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_PAUSED:
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_PLAYING_TO_PAUSED:
|
|
ret = GST_STATE_CHANGE_NO_PREROLL;
|
|
break;
|
|
case GST_STATE_CHANGE_READY_TO_NULL:
|
|
gst_rtp_src_stop (self);
|
|
break;
|
|
default:
|
|
break;
|
|
}
|
|
|
|
return ret;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_src_init (GstRtpSrc * self)
|
|
{
|
|
gchar name[48];
|
|
const gchar *missing_plugin = NULL;
|
|
|
|
self->rtpbin = NULL;
|
|
self->rtp_src = NULL;
|
|
self->rtcp_src = NULL;
|
|
self->rtcp_sink = NULL;
|
|
|
|
self->uri = gst_uri_from_string (DEFAULT_PROP_URI);
|
|
self->ttl = DEFAULT_PROP_TTL;
|
|
self->ttl_mc = DEFAULT_PROP_TTL_MC;
|
|
self->encoding_name = DEFAULT_PROP_ENCODING_NAME;
|
|
|
|
GST_OBJECT_FLAG_SET (GST_OBJECT (self), GST_ELEMENT_FLAG_SOURCE);
|
|
gst_bin_set_suppressed_flags (GST_BIN (self),
|
|
GST_ELEMENT_FLAG_SOURCE | GST_ELEMENT_FLAG_SINK);
|
|
|
|
g_mutex_init (&self->lock);
|
|
|
|
/* Construct the RTP receiver pipeline.
|
|
*
|
|
* udpsrc -> [recv_rtp_sink_%u] -------- [recv_rtp_src_%u_%u_%u]
|
|
* | rtpbin |
|
|
* udpsrc -> [recv_rtcp_sink_%u] -------- [send_rtcp_src_%u] -> udpsink
|
|
*
|
|
* This pipeline is fixed for now, note that optionally an FEC stream could
|
|
* be added later.
|
|
*/
|
|
|
|
self->rtpbin = gst_element_factory_make ("rtpbin", NULL);
|
|
if (self->rtpbin == NULL) {
|
|
missing_plugin = "rtpmanager";
|
|
goto missing_plugin;
|
|
}
|
|
|
|
gst_bin_add (GST_BIN (self), self->rtpbin);
|
|
|
|
/* Add rtpbin callbacks to monitor the operation of rtpbin */
|
|
g_signal_connect (self->rtpbin, "pad-added",
|
|
G_CALLBACK (gst_rtp_src_rtpbin_pad_added_cb), self);
|
|
g_signal_connect (self->rtpbin, "pad-removed",
|
|
G_CALLBACK (gst_rtp_src_rtpbin_pad_removed_cb), self);
|
|
g_signal_connect (self->rtpbin, "request-pt-map",
|
|
G_CALLBACK (gst_rtp_src_rtpbin_request_pt_map_cb), self);
|
|
g_signal_connect (self->rtpbin, "on-new-ssrc",
|
|
G_CALLBACK (gst_rtp_src_rtpbin_on_new_ssrc_cb), self);
|
|
g_signal_connect (self->rtpbin, "on-ssrc-collision",
|
|
G_CALLBACK (gst_rtp_src_rtpbin_on_ssrc_collision_cb), self);
|
|
|
|
self->rtp_src = gst_element_factory_make ("udpsrc", NULL);
|
|
if (self->rtp_src == NULL) {
|
|
missing_plugin = "udp";
|
|
goto missing_plugin;
|
|
}
|
|
|
|
self->rtcp_src = gst_element_factory_make ("udpsrc", NULL);
|
|
if (self->rtcp_src == NULL) {
|
|
missing_plugin = "udp";
|
|
goto missing_plugin;
|
|
}
|
|
|
|
self->rtcp_sink = gst_element_factory_make ("dynudpsink", NULL);
|
|
if (self->rtcp_sink == NULL) {
|
|
missing_plugin = "udp";
|
|
goto missing_plugin;
|
|
}
|
|
|
|
/* Add elements as needed, since udpsrc/udpsink for RTCP share a socket,
|
|
* not all at the same moment */
|
|
gst_bin_add (GST_BIN (self), self->rtp_src);
|
|
gst_bin_add (GST_BIN (self), self->rtcp_src);
|
|
gst_bin_add (GST_BIN (self), self->rtcp_sink);
|
|
|
|
g_object_set (self->rtcp_sink, "sync", FALSE, "async", FALSE, NULL);
|
|
gst_element_set_locked_state (self->rtcp_sink, TRUE);
|
|
|
|
/* pads are all named */
|
|
g_snprintf (name, 48, "recv_rtp_sink_%u", GST_ELEMENT (self)->numpads);
|
|
gst_element_link_pads (self->rtp_src, "src", self->rtpbin, name);
|
|
g_snprintf (name, 48, "recv_rtcp_sink_%u", GST_ELEMENT (self)->numpads);
|
|
gst_element_link_pads (self->rtcp_src, "src", self->rtpbin, name);
|
|
g_snprintf (name, 48, "send_rtcp_src_%u", GST_ELEMENT (self)->numpads);
|
|
gst_element_link_pads (self->rtpbin, name, self->rtcp_sink, "sink");
|
|
|
|
if (missing_plugin == NULL)
|
|
return;
|
|
|
|
missing_plugin:
|
|
{
|
|
GST_ERROR_OBJECT (self, "'%s' plugin is missing.", missing_plugin);
|
|
}
|
|
}
|
|
|
|
static GstURIType
|
|
gst_rtp_src_uri_get_type (GType type)
|
|
{
|
|
return GST_URI_SRC;
|
|
}
|
|
|
|
static const gchar *const *
|
|
gst_rtp_src_uri_get_protocols (GType type)
|
|
{
|
|
static const gchar *protocols[] = { (char *) "rtp", NULL };
|
|
|
|
return protocols;
|
|
}
|
|
|
|
static gchar *
|
|
gst_rtp_src_uri_get_uri (GstURIHandler * handler)
|
|
{
|
|
GstRtpSrc *self = (GstRtpSrc *) handler;
|
|
|
|
return gst_uri_to_string (self->uri);
|
|
}
|
|
|
|
static gboolean
|
|
gst_rtp_src_uri_set_uri (GstURIHandler * handler, const gchar * uri,
|
|
GError ** error)
|
|
{
|
|
GstRtpSrc *self = (GstRtpSrc *) handler;
|
|
|
|
g_object_set (G_OBJECT (self), "uri", uri, NULL);
|
|
|
|
return TRUE;
|
|
}
|
|
|
|
static void
|
|
gst_rtp_src_uri_handler_init (gpointer g_iface, gpointer iface_data)
|
|
{
|
|
GstURIHandlerInterface *iface = (GstURIHandlerInterface *) g_iface;
|
|
|
|
iface->get_type = gst_rtp_src_uri_get_type;
|
|
iface->get_protocols = gst_rtp_src_uri_get_protocols;
|
|
iface->get_uri = gst_rtp_src_uri_get_uri;
|
|
iface->set_uri = gst_rtp_src_uri_set_uri;
|
|
}
|
|
|
|
/* ex: set tabstop=2 shiftwidth=2 expandtab: */
|