Jetson Inference: gstWebRTC Class Reference

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gstWebRTC Class Reference Utilities Library (jetson-utils) » Codec

Static class for common WebRTC utility functions used with GStreamer. More...

#include <gstWebRTC.h>

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Classes

| | struct | PeerContext | | | GStreamer-specific context for each WebRTCPeer. More...
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Static Public Member Functions

| | static void | onNegotiationNeeded (GstElement *webrtcbin, void *user_data) | | | Callback for handling webrtcbin "on-negotation-needed" signal. More...
| | | | static void | onCreateOffer (GstPromise *promise, void *user_data) | | | Callback for handling webrtcbin "create-offer" signal. More...
| | | | static void | onIceCandidate (GstElement *webrtcbin, uint32_t mline_index, char *candidate, void *user_data) | | | Callback for handling webrtcbin "on-ice-candidate" signal. More...
| | | | static void | onWebsocketMessage (WebRTCPeer *peer, const char *message, size_t message_size, void *user_data) | | | Handle incoming websocket messages from the client. More...
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Detailed Description

Static class for common WebRTC utility functions used with GStreamer.

This gets used internally by gstEncoder/gstDecoder for handling WebRTC streams.

Member Function Documentation

◆onCreateOffer()

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| static void gstWebRTC::onCreateOffer | ( | GstPromise * | promise, | | | | void * | user_data | | | ) | | |

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Callback for handling webrtcbin "create-offer" signal.

This sends an SDP offer to the client.

◆onIceCandidate()

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| static void gstWebRTC::onIceCandidate | ( | GstElement * | webrtcbin, | | | | uint32_t | mline_index, | | | | char * | candidate, | | | | void * | user_data | | | ) | | |

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Callback for handling webrtcbin "on-ice-candidate" signal.

This send an ICE candidate to the client.

◆onNegotiationNeeded()

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| static void gstWebRTC::onNegotiationNeeded | ( | GstElement * | webrtcbin, | | | | void * | user_data | | | ) | | |

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Callback for handling webrtcbin "on-negotation-needed" signal.

It's expected that user_data is set to a WebRTCPeer instance.

◆onWebsocketMessage()

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| static void gstWebRTC::onWebsocketMessage | ( | WebRTCPeer * | peer, | | | | const char * | message, | | | | size_t | message_size, | | | | void * | user_data | | | ) | | |

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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.

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The documentation for this class was generated from the following file:

• jetson-utils/gstWebRTC.h

• gstWebRTC

• Generated on Fri Mar 17 2023 14:29:30 for Jetson Inference by 1.8.17