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gstWebRTC.h
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gstWebRTC.h
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/*
* Copyright (c) 2022, NVIDIA CORPORATION. All rights reserved.
*
* Permission is hereby granted, free of charge, to any person obtaining a
* copy of this software and associated documentation files (the "Software"),
* to deal in the Software without restriction, including without limitation
* the rights to use, copy, modify, merge, publish, distribute, sublicense,
* and/or sell copies of the Software, and to permit persons to whom the
* Software is furnished to do so, subject to the following conditions:
*
* The above copyright notice and this permission notice shall be included in
* all copies or substantial portions of the Software.
*
* THE SOFTWARE IS PROVIDED "AS IS", WITHOUT WARRANTY OF ANY KIND, EXPRESS OR
* IMPLIED, INCLUDING BUT NOT LIMITED TO THE WARRANTIES OF MERCHANTABILITY,
* FITNESS FOR A PARTICULAR PURPOSE AND NONINFRINGEMENT. IN NO EVENT SHALL
* THE AUTHORS OR COPYRIGHT HOLDERS BE LIABLE FOR ANY CLAIM, DAMAGES OR OTHER
* LIABILITY, WHETHER IN AN ACTION OF CONTRACT, TORT OR OTHERWISE, ARISING
* FROM, OUT OF OR IN CONNECTION WITH THE SOFTWARE OR THE USE OR OTHER
* DEALINGS IN THE SOFTWARE.
*/
#ifndef __GSTREAMER_WEBRTC_H__
#define __GSTREAMER_WEBRTC_H__
#include "WebRTCServer.h"
#include "gstUtility.h"
#define GST_USE_UNSTABLE_API
#include <gst/webrtc/webrtc.h>
/**
* Static class for common WebRTC utility functions used with GStreamer.
* This gets used internally by gstEncoder/gstDecoder for handling WebRTC streams.
* @ingroup codec
*/
class gstWebRTC
{
public:
/**
* GStreamer-specific context for each WebRTCPeer
*/
struct PeerContext
{
PeerContext() { webrtcbin = NULL; queue = NULL; }
GstElement* webrtcbin; // used by gstEncoder + gstDecoder
GstElement* queue; // used by gstEncoder only
};
/**
* Callback for handling webrtcbin "on-negotation-needed" signal.
* It's expected that user_data is set to a WebRTCPeer instance.
*/
static void onNegotiationNeeded( GstElement* webrtcbin, void* user_data );
/**
* Callback for handling webrtcbin "create-offer" signal.
* This sends an SDP offer to the client.
*/
static void onCreateOffer( GstPromise* promise, void* user_data );
/**
* Callback for handling webrtcbin "on-ice-candidate" signal.
* This send an ICE candidate to the client.
*/
static void onIceCandidate( GstElement* webrtcbin, uint32_t mline_index, char* candidate, void* user_data );
/**
* Handle incoming websocket messages from the client.
* This only handles SDP/ICE messages - it's expected that the caller will
* handle new peer connecting/closing messages.
*/
static void onWebsocketMessage( WebRTCPeer* peer, const char* message, size_t message_size, void* user_data );
};
#endif