#include <gst/gst.h>
#include <gst/sdp/sdp.h>
#include <gst/app/app.h>
#include <gst/webrtc/webrtc.h>
#include <libsoup/soup.h>
#include <json-glib/json-glib.h>
#include <string.h>
#include <ros/ros.h>
#include <image_transport/image_transport.h>
#include <sensor_msgs/image_encodings.h>
#include <thread>
#include <std_msgs/String.h>

Include dependency graph for webrtc.cpp:

Macros
#define	GST_USE_UNSTABLE_API

#define	RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="

#define	RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload="

#define	STUN_SERVER " stun-server=stun://stun.l.google.com:19302 "

Enumerations
enum	AppState { APP_STATE_UNKNOWN = 0, APP_STATE_ERROR = 1, SERVER_CONNECTING = 1000, SERVER_CONNECTION_ERROR, SERVER_CONNECTED, SERVER_REGISTERING = 2000, SERVER_REGISTRATION_ERROR, SERVER_REGISTERED, SERVER_CLOSED, PEER_CONNECTING = 3000, PEER_CONNECTION_ERROR, PEER_CONNECTED, PEER_CALL_NEGOTIATING = 4000, PEER_CALL_STARTED, PEER_CALL_STOPPING, PEER_CALL_STOPPED, PEER_CALL_ERROR }

Functions
static gboolean	check_plugins (void)

static gboolean	cleanup_and_quit_loop (const gchar *msg, AppState state)

static void	connect_to_websocket_server_async (void)

static gchar *	get_string_from_json_object (JsonObject *object)

GstCaps *	gst_caps_new_from_image (const sensor_msgs::Image::ConstPtr &msg)

GstSample *	gst_sample_new_from_image (const sensor_msgs::Image::ConstPtr &msg)

static void	handle_media_stream (GstPad pad, GstElement pipe, const char convert_name, const char sink_name)

void	image_cb (const sensor_msgs::Image::ConstPtr &msg)

int	main (int argc, char **argv)

static void	on_ice_gathering_state_notify (GstElement webrtcbin, GParamSpec pspec, gpointer user_data)

static void	on_incoming_decodebin_stream (GstElement decodebin, GstPad pad, GstElement *pipe)

static void	on_incoming_stream (GstElement webrtc, GstPad pad, GstElement *pipe)

static void	on_negotiation_needed (GstElement *element, gpointer user_data)

static void	on_offer_created (GstPromise *promise, gpointer user_data)

static void	on_server_closed (SoupWebsocketConnection *conn G_GNUC_UNUSED, gpointer user_data G_GNUC_UNUSED)

static void	on_server_connected (SoupSession session, GAsyncResult res, SoupMessage *msg)

static void	on_server_message (SoupWebsocketConnection conn, SoupWebsocketDataType type, GBytes message, gpointer user_data)

static gboolean	register_with_server (void)

static void	send_ice_candidate_message (GstElement webrtc G_GNUC_UNUSED, guint mlineindex, gchar candidate, gpointer user_data G_GNUC_UNUSED)

static void	send_sdp_to_peer (GstWebRTCSessionDescription *desc)

static gboolean	setup_call (void)

static gboolean	start_pipeline (void)

void	startLoopAndConnect ()

Variables
static AppState	app_state = APP_STATE_UNKNOWN

static GstElement *	appsrc_

static const gchar *	audio_device = nullptr

static gboolean	disable_ssl = FALSE

static GOptionEntry	entries []

image_transport::Subscriber	image_sub_

std::unique_ptr< image_transport::ImageTransport >	image_transport_

static GMainLoop *	loop

std::thread	loop_thread_

static const int	MAXRETRIES = 5

static const gchar *	peer_id = nullptr

static GstElement *	pipe1

static GObject *	receive_channel

static int	retries = 0

static GObject *	send_channel

static const gchar *	server_url = "wss://127.0.0.1:8443"

gboolean	serverConnected = FALSE

static GCancellable *	serverConnection

gboolean	serverConnectStarted = FALSE

static const gchar *	turnserver_login = nullptr

static const gchar *	turnserver_port = nullptr

static const gchar *	video_bitrate = nullptr

static const gchar *	video_height = nullptr

static const gchar *	video_width = nullptr

static GstElement *	webrtc1

static ros::Publisher	webrtcstatus_pub

static SoupWebsocketConnection *	ws_conn = nullptr

Macro Definition Documentation

#define GST_USE_UNSTABLE_API

Definition at line 10 of file webrtc.cpp.

#define RTP_CAPS_OPUS "application/x-rtp,media=audio,encoding-name=OPUS,payload="

Definition at line 343 of file webrtc.cpp.

#define RTP_CAPS_VP8 "application/x-rtp,media=video,encoding-name=VP8,payload="

Definition at line 344 of file webrtc.cpp.

#define STUN_SERVER " stun-server=stun://stun.l.google.com:19302 "

Definition at line 342 of file webrtc.cpp.

Enumeration Type Documentation

enum AppState

Enumerator
APP_STATE_UNKNOWN
APP_STATE_ERROR
SERVER_CONNECTING
SERVER_CONNECTION_ERROR
SERVER_CONNECTED
SERVER_REGISTERING
SERVER_REGISTRATION_ERROR
SERVER_REGISTERED
SERVER_CLOSED
PEER_CONNECTING
PEER_CONNECTION_ERROR
PEER_CONNECTED
PEER_CALL_NEGOTIATING
PEER_CALL_STARTED
PEER_CALL_STOPPING
PEER_CALL_STOPPED
PEER_CALL_ERROR

Definition at line 25 of file webrtc.cpp.

Function Documentation

static gboolean check_plugins ( void )

static

Definition at line 747 of file webrtc.cpp.

static gboolean cleanup_and_quit_loop	(	const gchar *	msg,
		AppState	state
	)

static

Definition at line 90 of file webrtc.cpp.

static void connect_to_websocket_server_async ( void )

static

Definition at line 714 of file webrtc.cpp.

static gchar* get_string_from_json_object ( JsonObject * object )

static

Definition at line 118 of file webrtc.cpp.

GstCaps* gst_caps_new_from_image ( const sensor_msgs::Image::ConstPtr & msg )

Definition at line 770 of file webrtc.cpp.

GstSample* gst_sample_new_from_image ( const sensor_msgs::Image::ConstPtr & msg )

Definition at line 805 of file webrtc.cpp.

static void handle_media_stream	(	GstPad *	pad,
		GstElement *	pipe,
		const char *	convert_name,
		const char *	sink_name
	)

static

Definition at line 137 of file webrtc.cpp.

void image_cb ( const sensor_msgs::Image::ConstPtr & msg )

Definition at line 846 of file webrtc.cpp.

int main	(	int	argc,
		char **	argv
	)

Definition at line 865 of file webrtc.cpp.

static void on_ice_gathering_state_notify	(	GstElement *	webrtcbin,
		GParamSpec *	pspec,
		gpointer	user_data
	)

static

Definition at line 320 of file webrtc.cpp.

static void on_incoming_decodebin_stream	(	GstElement *	decodebin,
		GstPad *	pad,
		GstElement *	pipe
	)

static

Definition at line 179 of file webrtc.cpp.

static void on_incoming_stream	(	GstElement *	webrtc,
		GstPad *	pad,
		GstElement *	pipe
	)

static

Definition at line 204 of file webrtc.cpp.

static void on_negotiation_needed	(	GstElement *	element,
		gpointer	user_data
	)

static

Definition at line 310 of file webrtc.cpp.

static void on_offer_created	(	GstPromise *	promise,
		gpointer	user_data
	)

static

Definition at line 286 of file webrtc.cpp.

static void on_server_closed	(	SoupWebsocketConnection *conn	G_GNUC_UNUSED,
		gpointer user_data	G_GNUC_UNUSED
	)

static

Definition at line 511 of file webrtc.cpp.

static void on_server_connected	(	SoupSession *	session,
		GAsyncResult *	res,
		SoupMessage *	msg
	)

static

Definition at line 680 of file webrtc.cpp.

static void on_server_message	(	SoupWebsocketConnection *	conn,
		SoupWebsocketDataType	type,
		GBytes *	message,
		gpointer	user_data
	)

static

Definition at line 525 of file webrtc.cpp.

static gboolean register_with_server ( void )

static

Definition at line 488 of file webrtc.cpp.

static void send_ice_candidate_message	(	GstElement *webrtc	G_GNUC_UNUSED,
		guint	mlineindex,
		gchar *	candidate,
		gpointer user_data	G_GNUC_UNUSED
	)

static

Definition at line 224 of file webrtc.cpp.

static void send_sdp_to_peer ( GstWebRTCSessionDescription * desc )

static

Definition at line 248 of file webrtc.cpp.

static gboolean setup_call ( void )

static

Definition at line 468 of file webrtc.cpp.

static gboolean start_pipeline ( void )

static

Definition at line 346 of file webrtc.cpp.

void startLoopAndConnect ( )

Definition at line 829 of file webrtc.cpp.

Variable Documentation

AppState app_state = APP_STATE_UNKNOWN

static

Definition at line 53 of file webrtc.cpp.

GstElement * appsrc_

static

Definition at line 49 of file webrtc.cpp.

const gchar* audio_device = nullptr

static

Definition at line 67 of file webrtc.cpp.

gboolean disable_ssl = FALSE

static

Definition at line 56 of file webrtc.cpp.

GOptionEntry entries[]

static

Initial value:

=
{
  { "peer-id", 0, 0, G_OPTION_ARG_STRING, &peer_id, "String ID of the peer to connect to", "ID" },
  { "server", 0, 0, G_OPTION_ARG_STRING, &server_url, "Signalling server to connect to", "URL" },
  { "disable-ssl", 0, 0, G_OPTION_ARG_NONE, &disable_ssl, "Disable ssl", NULL },
  { "audio-device", 0, 0, G_OPTION_ARG_STRING, &audio_device, "String of the audio device number", "ID" },
  { "video-width", 0, 0, G_OPTION_ARG_STRING, &video_width, "video width pixels", NULL },
  { "video-height", 0, 0, G_OPTION_ARG_STRING, &video_height, "video height pixels", NULL },
  { "video-bitrate", 0, 0, G_OPTION_ARG_STRING, &video_bitrate, "target bitrate kbps", NULL },
  { "turnserver-port", 0, 0, G_OPTION_ARG_STRING, &turnserver_port, "TURN server port number", NULL },
  { "turnserver-login", 0, 0, G_OPTION_ARG_STRING, &turnserver_login, "TURN server user:pass", NULL },
  { NULL },
}

Definition at line 75 of file webrtc.cpp.

image_transport::Subscriber image_sub_

Definition at line 60 of file webrtc.cpp.

std::unique_ptr<image_transport::ImageTransport> image_transport_

Definition at line 59 of file webrtc.cpp.

GMainLoop* loop

static

Definition at line 48 of file webrtc.cpp.

std::thread loop_thread_

Definition at line 62 of file webrtc.cpp.

const int MAXRETRIES = 5

static

Definition at line 64 of file webrtc.cpp.

const gchar* peer_id = nullptr

static

Definition at line 54 of file webrtc.cpp.

GstElement* pipe1

static

Definition at line 49 of file webrtc.cpp.

GObject * receive_channel

static

Definition at line 50 of file webrtc.cpp.

int retries = 0

static

Definition at line 63 of file webrtc.cpp.

GObject* send_channel

static

Definition at line 50 of file webrtc.cpp.

const gchar* server_url = "wss://127.0.0.1:8443"

static

Definition at line 55 of file webrtc.cpp.

gboolean serverConnected = FALSE

Definition at line 65 of file webrtc.cpp.

GCancellable* serverConnection

static

Definition at line 61 of file webrtc.cpp.

gboolean serverConnectStarted = FALSE

Definition at line 66 of file webrtc.cpp.

const gchar* turnserver_login = nullptr

static

Definition at line 71 of file webrtc.cpp.

const gchar* turnserver_port = nullptr

static

Definition at line 72 of file webrtc.cpp.

const gchar* video_bitrate = nullptr

static

Definition at line 70 of file webrtc.cpp.

const gchar* video_height = nullptr

static

Definition at line 69 of file webrtc.cpp.

const gchar* video_width = nullptr

static

Definition at line 68 of file webrtc.cpp.

GstElement * webrtc1

static

Definition at line 49 of file webrtc.cpp.

ros::Publisher webrtcstatus_pub

static

Definition at line 73 of file webrtc.cpp.

SoupWebsocketConnection* ws_conn = nullptr

static

Definition at line 52 of file webrtc.cpp.

Macros

Enumerations

Functions

Variables

Macro Definition Documentation

Enumeration Type Documentation

Function Documentation

Variable Documentation