Add option for custom port and fix indentation
This commit is contained in:
George Kouros
parent
978484a774
commit
75290a64bc
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@ -1,4 +1,5 @@
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# Set up your streams to rtsp here.
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port: "8554"
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streams: # Cannot rename - must leave this as is.
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stream-x:
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@ -13,4 +14,4 @@ streams: # Cannot rename - must leave this as is.
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mountpoint: /back
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caps: video/x-raw,framerate=10/1,width=640,height=480
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bitrate: 500
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@ -2,25 +2,26 @@
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#define IMAGE_TO_RTSP_H
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namespace image2rtsp {
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class Image2RTSPNodelet : public nodelet::Nodelet {
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public:
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GstRTSPServer *rtsp_server;
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void onInit();
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void url_connected(std::string url);
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void url_disconnected(std::string url);
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void print_info(char *s);
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void print_error(char *s);
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class Image2RTSPNodelet : public nodelet::Nodelet {
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public:
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GstRTSPServer *rtsp_server;
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void onInit();
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void url_connected(std::string url);
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void url_disconnected(std::string url);
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void print_info(char *s);
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void print_error(char *s);
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private:
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std::map<std::string, ros::Subscriber> subs;
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std::map<std::string, GstAppSrc*> appsrc;
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std::map<std::string, int> num_of_clients;
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GstCaps* gst_caps_new_from_image(const sensor_msgs::Image::ConstPtr &msg);
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void imageCallback(const sensor_msgs::Image::ConstPtr& msg, const std::string& topic);
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void video_mainloop_start();
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void rtsp_server_add_url(const char *url, const char *sPipeline, GstElement **appsrc);
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GstRTSPServer *rtsp_server_create();
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};
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private:
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std::string port;
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std::map<std::string, ros::Subscriber> subs;
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std::map<std::string, GstAppSrc*> appsrc;
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std::map<std::string, int> num_of_clients;
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GstCaps* gst_caps_new_from_image(const sensor_msgs::Image::ConstPtr &msg);
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void imageCallback(const sensor_msgs::Image::ConstPtr& msg, const std::string& topic);
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void video_mainloop_start();
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void rtsp_server_add_url(const char *url, const char *sPipeline, GstElement **appsrc);
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GstRTSPServer *rtsp_server_create(const std::string& port);
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};
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}
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#endif
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@ -3,9 +3,9 @@
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<!-- Start the RTSP server -->
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<node pkg="nodelet" type="nodelet" name="standalone_nodelet" args="manager" output="screen"/>
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<node pkg="nodelet" type="nodelet" name="Image2RTSPNodelet" args="load image2rtsp/Image2RTSPNodelet standalone_nodelet" output="screen">
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<node pkg="nodelet" type="nodelet" name="Image2RTSPNodelet" args="load image2rtsp/Image2RTSPNodelet standalone_nodelet" output="screen">
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<!-- Read the stream setup file -->
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<rosparam command="load" file="$(find ros_rtsp)/config/stream_setup.yaml" />
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</node>
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</node>
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</launch>
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@ -1,7 +1,7 @@
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<library path="lib/libimage_to_rtsp_nodelet">
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<class name="image2rtsp/Image2RTSPNodelet" type="image2rtsp::Image2RTSPNodelet" base_class_type="nodelet::Nodelet">
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<description>
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Nodelet to transfert the /camera/image topic from openni2_camera over rtsp.
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</description>
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</class>
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<class name="image2rtsp/Image2RTSPNodelet" type="image2rtsp::Image2RTSPNodelet" base_class_type="nodelet::Nodelet">
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<description>
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Nodelet to transfert the /camera/image topic from openni2_camera over rtsp.
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</description>
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</class>
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</library>
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@ -16,191 +16,191 @@ using namespace image2rtsp;
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void Image2RTSPNodelet::onInit() {
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string pipeline, mountpoint, bitrate, caps;
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string pipeline_tail = " key-int-max=30 ! video/x-h264, profile=baseline ! rtph264pay name=pay0 pt=96 )"; // Gets completed based on rosparams below
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string pipeline, mountpoint, bitrate, caps;
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string pipeline_tail = " key-int-max=30 ! video/x-h264, profile=baseline ! rtph264pay name=pay0 pt=96 )"; // Gets completed based on rosparams below
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NODELET_DEBUG("Initializing image2rtsp nodelet...");
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NODELET_DEBUG("Initializing image2rtsp nodelet...");
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if (getenv((char*)"GST_DEBUG") == NULL) {
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// set GST_DEBUG to warning if unset
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putenv((char*)"GST_DEBUG=*:1");
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}
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if (getenv((char*)"GST_DEBUG") == NULL) {
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// set GST_DEBUG to warning if unset
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putenv((char*)"GST_DEBUG=*:1");
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}
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ros::NodeHandle& nh = getPrivateNodeHandle();
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ros::NodeHandle& nh = getPrivateNodeHandle();
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video_mainloop_start();
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rtsp_server = rtsp_server_create();
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// Get the parameters from the rosparam server
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XmlRpc::XmlRpcValue streams;
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nh.getParam("streams", streams);
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ROS_ASSERT(streams.getType() == XmlRpc::XmlRpcValue::TypeStruct);
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ROS_DEBUG("Number of RTSP streams: %d", streams.size());
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nh.getParam("port", this->port);
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// Get the parameters from the rosparam server
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XmlRpc::XmlRpcValue streams;
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nh.getParam("streams", streams);
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ROS_ASSERT(streams.getType() == XmlRpc::XmlRpcValue::TypeStruct);
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video_mainloop_start();
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rtsp_server = rtsp_server_create(port);
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ROS_DEBUG("Number of RTSP streams: %d", streams.size());
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// Go through and parse each stream
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for(XmlRpc::XmlRpcValue::ValueStruct::const_iterator it = streams.begin(); it != streams.end(); ++it)
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{
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XmlRpc::XmlRpcValue stream = streams[it->first];
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ROS_DEBUG_STREAM("Found stream: " << (std::string)(it->first) << " ==> " << stream);
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// Go through and parse each stream
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for(XmlRpc::XmlRpcValue::ValueStruct::const_iterator it = streams.begin(); it != streams.end(); ++it)
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{
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XmlRpc::XmlRpcValue stream = streams[it->first];
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ROS_DEBUG_STREAM("Found stream: " << (std::string)(it->first) << " ==> " << stream);
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// Convert to string for ease of use
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mountpoint = static_cast<std::string>(stream["mountpoint"]);
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bitrate = std::to_string(static_cast<int>(stream["bitrate"]));
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// Convert to string for ease of use
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mountpoint = static_cast<std::string>(stream["mountpoint"]);
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bitrate = std::to_string(static_cast<int>(stream["bitrate"]));
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// uvc camera?
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if (stream["type"]=="cam")
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{
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pipeline = "( " + static_cast<std::string>(stream["source"]) + " ! x264enc tune=zerolatency bitrate=" + bitrate + pipeline_tail;
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// uvc camera?
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if (stream["type"]=="cam")
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{
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pipeline = "( " + static_cast<std::string>(stream["source"]) + " ! x264enc tune=zerolatency bitrate=" + bitrate + pipeline_tail;
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rtsp_server_add_url(mountpoint.c_str(), pipeline.c_str(), NULL);
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}
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// ROS Image topic?
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else if (stream["type"]=="topic")
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{
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/* Keep record of number of clients connected to each topic.
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* so we know to stop subscribing when no-one is connected. */
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num_of_clients[mountpoint] = 0;
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appsrc[mountpoint] = NULL;
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caps = static_cast<std::string>(stream["caps"]);
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rtsp_server_add_url(mountpoint.c_str(), pipeline.c_str(), NULL);
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}
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// ROS Image topic?
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else if (stream["type"]=="topic")
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{
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/* Keep record of number of clients connected to each topic.
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* so we know to stop subscribing when no-one is connected. */
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num_of_clients[mountpoint] = 0;
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appsrc[mountpoint] = NULL;
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caps = static_cast<std::string>(stream["caps"]);
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// Setup the full pipeline
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pipeline = "( appsrc name=imagesrc do-timestamp=true min-latency=0 max-latency=0 max-bytes=1000 is-live=true ! videoconvert ! videoscale ! " + caps + " ! x264enc tune=zerolatency bitrate=" + bitrate + pipeline_tail;
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// Setup the full pipeline
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pipeline = "( appsrc name=imagesrc do-timestamp=true min-latency=0 max-latency=0 max-bytes=1000 is-live=true ! videoconvert ! videoscale ! " + caps + " ! x264enc tune=zerolatency bitrate=" + bitrate + pipeline_tail;
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// Add the pipeline to the rtsp server
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rtsp_server_add_url(mountpoint.c_str(), pipeline.c_str(), (GstElement **)&(appsrc[mountpoint]));
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}
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else
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{
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ROS_ERROR("Undefined stream type. Check your stream_setup.yaml file.");
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}
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NODELET_INFO("Stream available at rtsp://%s:8554%s", gst_rtsp_server_get_address(rtsp_server), mountpoint.c_str());
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}
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// Add the pipeline to the rtsp server
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rtsp_server_add_url(mountpoint.c_str(), pipeline.c_str(), (GstElement **)&(appsrc[mountpoint]));
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}
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else
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{
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ROS_ERROR("Undefined stream type. Check your stream_setup.yaml file.");
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}
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NODELET_INFO("Stream available at rtsp://%s:%s%s", gst_rtsp_server_get_address(rtsp_server), port.c_str(), mountpoint.c_str());
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}
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}
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/* Modified from https://github.com/ProjectArtemis/gst_video_server/blob/master/src/server_nodelet.cpp */
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GstCaps* Image2RTSPNodelet::gst_caps_new_from_image(const sensor_msgs::Image::ConstPtr &msg)
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{
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// http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/section-types-definitions.html
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static const ros::M_string known_formats = {{
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{sensor_msgs::image_encodings::RGB8, "RGB"},
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{sensor_msgs::image_encodings::RGB16, "RGB16"},
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{sensor_msgs::image_encodings::RGBA8, "RGBA"},
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{sensor_msgs::image_encodings::RGBA16, "RGBA16"},
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{sensor_msgs::image_encodings::BGR8, "BGR"},
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{sensor_msgs::image_encodings::BGR16, "BGR16"},
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{sensor_msgs::image_encodings::BGRA8, "BGRA"},
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{sensor_msgs::image_encodings::BGRA16, "BGRA16"},
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{sensor_msgs::image_encodings::MONO8, "GRAY8"},
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{sensor_msgs::image_encodings::MONO16, "GRAY16_LE"},
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}};
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// http://gstreamer.freedesktop.org/data/doc/gstreamer/head/pwg/html/section-types-definitions.html
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static const ros::M_string known_formats = {{
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{sensor_msgs::image_encodings::RGB8, "RGB"},
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{sensor_msgs::image_encodings::RGB16, "RGB16"},
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{sensor_msgs::image_encodings::RGBA8, "RGBA"},
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{sensor_msgs::image_encodings::RGBA16, "RGBA16"},
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{sensor_msgs::image_encodings::BGR8, "BGR"},
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{sensor_msgs::image_encodings::BGR16, "BGR16"},
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{sensor_msgs::image_encodings::BGRA8, "BGRA"},
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{sensor_msgs::image_encodings::BGRA16, "BGRA16"},
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{sensor_msgs::image_encodings::MONO8, "GRAY8"},
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{sensor_msgs::image_encodings::MONO16, "GRAY16_LE"},
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}};
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if (msg->is_bigendian) {
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ROS_ERROR("GST: big endian image format is not supported");
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return nullptr;
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}
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if (msg->is_bigendian) {
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ROS_ERROR("GST: big endian image format is not supported");
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return nullptr;
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}
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auto format = known_formats.find(msg->encoding);
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if (format == known_formats.end()) {
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ROS_ERROR("GST: image format '%s' unknown", msg->encoding.c_str());
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return nullptr;
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}
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auto format = known_formats.find(msg->encoding);
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if (format == known_formats.end()) {
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ROS_ERROR("GST: image format '%s' unknown", msg->encoding.c_str());
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return nullptr;
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}
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return gst_caps_new_simple("video/x-raw",
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"format", G_TYPE_STRING, format->second.c_str(),
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"width", G_TYPE_INT, msg->width,
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"height", G_TYPE_INT, msg->height,
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"framerate", GST_TYPE_FRACTION, 10, 1,
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nullptr);
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return gst_caps_new_simple("video/x-raw",
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"format", G_TYPE_STRING, format->second.c_str(),
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"width", G_TYPE_INT, msg->width,
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"height", G_TYPE_INT, msg->height,
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"framerate", GST_TYPE_FRACTION, 10, 1,
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nullptr);
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}
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void Image2RTSPNodelet::imageCallback(const sensor_msgs::Image::ConstPtr& msg, const std::string& topic) {
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GstBuffer *buf;
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GstBuffer *buf;
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GstCaps *caps;
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char *gst_type, *gst_format=(char *)"";
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// g_print("Image encoding: %s\n", msg->encoding.c_str());
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if (appsrc[topic] != NULL) {
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// Set caps from message
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caps = gst_caps_new_from_image(msg);
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gst_app_src_set_caps(appsrc[topic], caps);
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GstCaps *caps;
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char *gst_type, *gst_format=(char *)"";
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// g_print("Image encoding: %s\n", msg->encoding.c_str());
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if (appsrc[topic] != NULL) {
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// Set caps from message
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caps = gst_caps_new_from_image(msg);
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gst_app_src_set_caps(appsrc[topic], caps);
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buf = gst_buffer_new_allocate(nullptr, msg->data.size(), nullptr);
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gst_buffer_fill(buf, 0, msg->data.data(), msg->data.size());
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GST_BUFFER_FLAG_SET(buf, GST_BUFFER_FLAG_LIVE);
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buf = gst_buffer_new_allocate(nullptr, msg->data.size(), nullptr);
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gst_buffer_fill(buf, 0, msg->data.data(), msg->data.size());
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GST_BUFFER_FLAG_SET(buf, GST_BUFFER_FLAG_LIVE);
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gst_app_src_push_buffer(appsrc[topic], buf);
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}
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gst_app_src_push_buffer(appsrc[topic], buf);
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}
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}
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void Image2RTSPNodelet::url_connected(string url) {
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std::string mountpoint, source, type;
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std::string mountpoint, source, type;
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NODELET_INFO("Client connected: %s", url.c_str());
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ros::NodeHandle& nh = getPrivateNodeHandle();
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NODELET_INFO("Client connected: %s", url.c_str());
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ros::NodeHandle& nh = getPrivateNodeHandle();
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// Get the parameters from the rosparam server
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XmlRpc::XmlRpcValue streams;
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nh.getParam("streams", streams);
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ROS_ASSERT(streams.getType() == XmlRpc::XmlRpcValue::TypeStruct);
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// Get the parameters from the rosparam server
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XmlRpc::XmlRpcValue streams;
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nh.getParam("streams", streams);
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ROS_ASSERT(streams.getType() == XmlRpc::XmlRpcValue::TypeStruct);
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// Go through and parse each stream
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for(XmlRpc::XmlRpcValue::ValueStruct::const_iterator it = streams.begin(); it != streams.end(); ++it)
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{
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XmlRpc::XmlRpcValue stream = streams[it->first];
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type = static_cast<std::string>(stream["type"]);
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mountpoint = static_cast<std::string>(stream["mountpoint"]);
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source = static_cast<std::string>(stream["source"]);
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// Go through and parse each stream
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for(XmlRpc::XmlRpcValue::ValueStruct::const_iterator it = streams.begin(); it != streams.end(); ++it)
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{
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XmlRpc::XmlRpcValue stream = streams[it->first];
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type = static_cast<std::string>(stream["type"]);
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mountpoint = static_cast<std::string>(stream["mountpoint"]);
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source = static_cast<std::string>(stream["source"]);
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// Check which stream the client has connected to
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if (type=="topic" && url==mountpoint) {
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// Check which stream the client has connected to
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if (type=="topic" && url==mountpoint) {
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if (num_of_clients[url]==0) {
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// Subscribe to the ROS topic
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subs[url] = nh.subscribe<sensor_msgs::Image>(source, 1, boost::bind(&Image2RTSPNodelet::imageCallback, this, _1, url));
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}
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num_of_clients[url]++;
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}
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}
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if (num_of_clients[url]==0) {
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// Subscribe to the ROS topic
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subs[url] = nh.subscribe<sensor_msgs::Image>(source, 1, boost::bind(&Image2RTSPNodelet::imageCallback, this, _1, url));
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}
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num_of_clients[url]++;
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}
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}
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}
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void Image2RTSPNodelet::url_disconnected(string url) {
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std::string mountpoint;
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std::string mountpoint;
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NODELET_INFO("Client disconnected: %s", url.c_str());
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ros::NodeHandle& nh = getPrivateNodeHandle();
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NODELET_INFO("Client disconnected: %s", url.c_str());
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ros::NodeHandle& nh = getPrivateNodeHandle();
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// Get the parameters from the rosparam server
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XmlRpc::XmlRpcValue streams;
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nh.getParam("streams", streams);
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ROS_ASSERT(streams.getType() == XmlRpc::XmlRpcValue::TypeStruct);
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// Get the parameters from the rosparam server
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XmlRpc::XmlRpcValue streams;
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nh.getParam("streams", streams);
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ROS_ASSERT(streams.getType() == XmlRpc::XmlRpcValue::TypeStruct);
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// Go through and parse each stream
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for(XmlRpc::XmlRpcValue::ValueStruct::const_iterator it = streams.begin(); it != streams.end(); ++it)
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{
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XmlRpc::XmlRpcValue stream = streams[it->first];
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mountpoint = static_cast<std::string>(stream["mountpoint"]);
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// Go through and parse each stream
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for(XmlRpc::XmlRpcValue::ValueStruct::const_iterator it = streams.begin(); it != streams.end(); ++it)
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{
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XmlRpc::XmlRpcValue stream = streams[it->first];
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mountpoint = static_cast<std::string>(stream["mountpoint"]);
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// Check which stream the client has disconnected from
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if (url==mountpoint) {
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if (num_of_clients[url] > 0) num_of_clients[url]--;
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if (num_of_clients[url] == 0) {
|
||||
// No-one else is connected. Stop the subscription.
|
||||
subs[url].shutdown();
|
||||
appsrc[url] = NULL;
|
||||
}
|
||||
}
|
||||
}
|
||||
// Check which stream the client has disconnected from
|
||||
if (url==mountpoint) {
|
||||
if (num_of_clients[url] > 0) num_of_clients[url]--;
|
||||
if (num_of_clients[url] == 0) {
|
||||
// No-one else is connected. Stop the subscription.
|
||||
subs[url].shutdown();
|
||||
appsrc[url] = NULL;
|
||||
}
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
void Image2RTSPNodelet::print_info(char *s) {
|
||||
NODELET_INFO("%s",s);
|
||||
NODELET_INFO("%s",s);
|
||||
}
|
||||
|
||||
void Image2RTSPNodelet::print_error(char *s) {
|
||||
NODELET_ERROR("%s",s);
|
||||
NODELET_ERROR("%s",s);
|
||||
}
|
||||
|
||||
PLUGINLIB_EXPORT_CLASS(image2rtsp::Image2RTSPNodelet, nodelet::Nodelet)
|
||||
|
|
|
|||
173
src/video.cpp
173
src/video.cpp
|
|
@ -16,147 +16,150 @@ using namespace image2rtsp;
|
|||
|
||||
|
||||
static void *mainloop(void *arg) {
|
||||
GMainLoop *loop = g_main_loop_new(NULL, FALSE);
|
||||
GMainLoop *loop = g_main_loop_new(NULL, FALSE);
|
||||
|
||||
g_main_loop_run(loop);
|
||||
g_main_loop_run(loop);
|
||||
|
||||
g_main_destroy(loop);
|
||||
return NULL;
|
||||
g_main_destroy(loop);
|
||||
return NULL;
|
||||
}
|
||||
|
||||
|
||||
void Image2RTSPNodelet::video_mainloop_start() {
|
||||
pthread_t tloop;
|
||||
pthread_t tloop;
|
||||
|
||||
gst_init(NULL, NULL);
|
||||
pthread_create(&tloop, NULL, &mainloop, NULL);
|
||||
gst_init(NULL, NULL);
|
||||
pthread_create(&tloop, NULL, &mainloop, NULL);
|
||||
}
|
||||
|
||||
|
||||
static void client_options(GstRTSPClient *client, GstRTSPContext *state, Image2RTSPNodelet *nodelet) {
|
||||
if (state->uri) {
|
||||
nodelet->url_connected(state->uri->abspath);
|
||||
}
|
||||
if (state->uri) {
|
||||
nodelet->url_connected(state->uri->abspath);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
static void client_teardown(GstRTSPClient *client, GstRTSPContext *state, Image2RTSPNodelet *nodelet) {
|
||||
if (state->uri) {
|
||||
nodelet->url_disconnected(state->uri->abspath);
|
||||
}
|
||||
if (state->uri) {
|
||||
nodelet->url_disconnected(state->uri->abspath);
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
static void new_client(GstRTSPServer *server, GstRTSPClient *client, Image2RTSPNodelet *nodelet) {
|
||||
nodelet->print_info((char *)"New RTSP client");
|
||||
g_signal_connect(client, "options-request", G_CALLBACK(client_options), nodelet);
|
||||
g_signal_connect(client, "teardown-request", G_CALLBACK(client_teardown), nodelet);
|
||||
nodelet->print_info((char *)"New RTSP client");
|
||||
g_signal_connect(client, "options-request", G_CALLBACK(client_options), nodelet);
|
||||
g_signal_connect(client, "teardown-request", G_CALLBACK(client_teardown), nodelet);
|
||||
}
|
||||
|
||||
/* this function is periodically run to clean up the expired sessions from the pool. */
|
||||
static gboolean session_cleanup(Image2RTSPNodelet *nodelet, gboolean ignored)
|
||||
{
|
||||
GstRTSPServer *server = nodelet->rtsp_server;
|
||||
GstRTSPSessionPool *pool;
|
||||
int num;
|
||||
GstRTSPServer *server = nodelet->rtsp_server;
|
||||
GstRTSPSessionPool *pool;
|
||||
int num;
|
||||
|
||||
pool = gst_rtsp_server_get_session_pool(server);
|
||||
num = gst_rtsp_session_pool_cleanup(pool);
|
||||
g_object_unref(pool);
|
||||
pool = gst_rtsp_server_get_session_pool(server);
|
||||
num = gst_rtsp_session_pool_cleanup(pool);
|
||||
g_object_unref(pool);
|
||||
|
||||
if (num > 0) {
|
||||
char s[32];
|
||||
snprintf(s, 32, (char *)"Sessions cleaned: %d", num);
|
||||
nodelet->print_info(s);
|
||||
}
|
||||
if (num > 0) {
|
||||
char s[32];
|
||||
snprintf(s, 32, (char *)"Sessions cleaned: %d", num);
|
||||
nodelet->print_info(s);
|
||||
}
|
||||
|
||||
return TRUE;
|
||||
return TRUE;
|
||||
}
|
||||
|
||||
GstRTSPServer *Image2RTSPNodelet::rtsp_server_create() {
|
||||
GstRTSPServer *server;
|
||||
GstRTSPServer *Image2RTSPNodelet::rtsp_server_create(const std::string& port) {
|
||||
GstRTSPServer *server;
|
||||
|
||||
/* create a server instance */
|
||||
server = gst_rtsp_server_new();
|
||||
/* create a server instance */
|
||||
server = gst_rtsp_server_new();
|
||||
|
||||
/* attach the server to the default maincontext */
|
||||
gst_rtsp_server_attach(server, NULL);
|
||||
// char *port = (char *) port;
|
||||
g_object_set(server, "service", port.c_str(), NULL);
|
||||
|
||||
g_signal_connect(server, "client-connected", G_CALLBACK(new_client), this);
|
||||
/* attach the server to the default maincontext */
|
||||
gst_rtsp_server_attach(server, NULL);
|
||||
|
||||
/* add a timeout for the session cleanup */
|
||||
g_timeout_add_seconds(2, (GSourceFunc)session_cleanup, this);
|
||||
g_signal_connect(server, "client-connected", G_CALLBACK(new_client), this);
|
||||
|
||||
return server;
|
||||
/* add a timeout for the session cleanup */
|
||||
g_timeout_add_seconds(2, (GSourceFunc)session_cleanup, this);
|
||||
|
||||
return server;
|
||||
}
|
||||
|
||||
|
||||
/* called when a new media pipeline is constructed. We can query the
|
||||
* pipeline and configure our appsrc */
|
||||
static void media_configure(GstRTSPMediaFactory *factory, GstRTSPMedia *media, GstElement **appsrc)
|
||||
{ if (appsrc) {
|
||||
GstElement *pipeline = gst_rtsp_media_get_element(media);
|
||||
{ if (appsrc) {
|
||||
GstElement *pipeline = gst_rtsp_media_get_element(media);
|
||||
|
||||
*appsrc = gst_bin_get_by_name(GST_BIN(pipeline), "imagesrc");
|
||||
*appsrc = gst_bin_get_by_name(GST_BIN(pipeline), "imagesrc");
|
||||
|
||||
/* this instructs appsrc that we will be dealing with timed buffer */
|
||||
gst_util_set_object_arg(G_OBJECT(*appsrc), "format", "time");
|
||||
/* this instructs appsrc that we will be dealing with timed buffer */
|
||||
gst_util_set_object_arg(G_OBJECT(*appsrc), "format", "time");
|
||||
|
||||
gst_object_unref(pipeline);
|
||||
}
|
||||
else
|
||||
{
|
||||
guint i, n_streams;
|
||||
n_streams = gst_rtsp_media_n_streams (media);
|
||||
gst_object_unref(pipeline);
|
||||
}
|
||||
else
|
||||
{
|
||||
guint i, n_streams;
|
||||
n_streams = gst_rtsp_media_n_streams (media);
|
||||
|
||||
for (i = 0; i < n_streams; i++) {
|
||||
GstRTSPAddressPool *pool;
|
||||
GstRTSPStream *stream;
|
||||
gchar *min, *max;
|
||||
for (i = 0; i < n_streams; i++) {
|
||||
GstRTSPAddressPool *pool;
|
||||
GstRTSPStream *stream;
|
||||
gchar *min, *max;
|
||||
|
||||
stream = gst_rtsp_media_get_stream (media, i);
|
||||
stream = gst_rtsp_media_get_stream (media, i);
|
||||
|
||||
/* make a new address pool */
|
||||
pool = gst_rtsp_address_pool_new ();
|
||||
/* make a new address pool */
|
||||
pool = gst_rtsp_address_pool_new ();
|
||||
|
||||
min = g_strdup_printf ("224.3.0.%d", (2 * i) + 1);
|
||||
max = g_strdup_printf ("224.3.0.%d", (2 * i) + 2);
|
||||
gst_rtsp_address_pool_add_range (pool, min, max,
|
||||
5000 + (10 * i), 5010 + (10 * i), 1);
|
||||
g_free (min);
|
||||
g_free (max);
|
||||
min = g_strdup_printf ("224.3.0.%d", (2 * i) + 1);
|
||||
max = g_strdup_printf ("224.3.0.%d", (2 * i) + 2);
|
||||
gst_rtsp_address_pool_add_range (pool, min, max,
|
||||
5000 + (10 * i), 5010 + (10 * i), 1);
|
||||
g_free (min);
|
||||
g_free (max);
|
||||
|
||||
gst_rtsp_stream_set_address_pool (stream, pool);
|
||||
g_object_unref (pool);
|
||||
}
|
||||
}
|
||||
gst_rtsp_stream_set_address_pool (stream, pool);
|
||||
g_object_unref (pool);
|
||||
}
|
||||
}
|
||||
}
|
||||
|
||||
|
||||
void Image2RTSPNodelet::rtsp_server_add_url(const char *url, const char *sPipeline, GstElement **appsrc) {
|
||||
GstRTSPMountPoints *mounts;
|
||||
GstRTSPMediaFactory *factory;
|
||||
GstRTSPMountPoints *mounts;
|
||||
GstRTSPMediaFactory *factory;
|
||||
|
||||
/* get the mount points for this server, every server has a default object
|
||||
* that be used to map uri mount points to media factories */
|
||||
mounts = gst_rtsp_server_get_mount_points(rtsp_server);
|
||||
/* get the mount points for this server, every server has a default object
|
||||
* that be used to map uri mount points to media factories */
|
||||
mounts = gst_rtsp_server_get_mount_points(rtsp_server);
|
||||
|
||||
/* make a media factory for a test stream. The default media factory can use
|
||||
* gst-launch syntax to create pipelines.
|
||||
* any launch line works as long as it contains elements named pay%d. Each
|
||||
* element with pay%d names will be a stream */
|
||||
factory = gst_rtsp_media_factory_new();
|
||||
gst_rtsp_media_factory_set_launch(factory, sPipeline);
|
||||
/* make a media factory for a test stream. The default media factory can use
|
||||
* gst-launch syntax to create pipelines.
|
||||
* any launch line works as long as it contains elements named pay%d. Each
|
||||
* element with pay%d names will be a stream */
|
||||
factory = gst_rtsp_media_factory_new();
|
||||
gst_rtsp_media_factory_set_launch(factory, sPipeline);
|
||||
|
||||
/* notify when our media is ready, This is called whenever someone asks for
|
||||
* the media and a new pipeline is created */
|
||||
g_signal_connect(factory, "media-configure", (GCallback)media_configure, appsrc);
|
||||
/* notify when our media is ready, This is called whenever someone asks for
|
||||
* the media and a new pipeline is created */
|
||||
g_signal_connect(factory, "media-configure", (GCallback)media_configure, appsrc);
|
||||
|
||||
gst_rtsp_media_factory_set_shared(factory, TRUE);
|
||||
gst_rtsp_media_factory_set_shared(factory, TRUE);
|
||||
|
||||
/* attach the factory to the url */
|
||||
gst_rtsp_mount_points_add_factory(mounts, url, factory);
|
||||
/* attach the factory to the url */
|
||||
gst_rtsp_mount_points_add_factory(mounts, url, factory);
|
||||
|
||||
/* don't need the ref to the mounts anymore */
|
||||
g_object_unref(mounts);
|
||||
/* don't need the ref to the mounts anymore */
|
||||
g_object_unref(mounts);
|
||||
}
|
||||
|
|
|
|||
Loading…
Reference in New Issue