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https://github.com/mollyim/webrtc.git
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fe9009be92
The gn files are simplified from what's currently in the RingRTC repo, looking ahead to when the RingRTC Rust and Java builds for Android aren't included in the WebRTC build.
74 lines
3.4 KiB
C++
74 lines
3.4 KiB
C++
/*
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* Copyright 2019-2021 Signal Messenger, LLC
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* SPDX-License-Identifier: AGPL-3.0-only
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*/
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#ifndef RFFI_API_PEER_CONNECTION_OBSERVER_INTF_H__
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#define RFFI_API_PEER_CONNECTION_OBSERVER_INTF_H__
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#include "api/peer_connection_interface.h"
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#include "rffi/api/rffi_defs.h"
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#include "rffi/api/media.h"
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#include "rffi/api/network.h"
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#include "rtc_base/network_constants.h"
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/**
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* Rust friendly wrapper around a custom class that implements the
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* webrtc::PeerConnectionObserver interface.
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*
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*/
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namespace webrtc {
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namespace rffi {
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class PeerConnectionObserverRffi;
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/* NetworkRoute structure passed between Rust and C++ */
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typedef struct {
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rtc::AdapterType local_adapter_type;
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rtc::AdapterType local_adapter_type_under_vpn;
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bool local_relayed;
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TransportProtocol local_relay_protocol;
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bool remote_relayed;
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} NetworkRoute;
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} // namespace rffi
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} // namespace webrtc
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/* Peer Connection Observer callback function pointers */
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typedef struct {
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// ICE events
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void (*onIceCandidate)(void* observer_borrowed, const RustIceCandidate* candidate_borrowed);
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void (*onIceCandidatesRemoved)(void* observer_borrowed, const webrtc::rffi::IpPort* addresses_borrowed, size_t);
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void (*onIceConnectionChange)(void* observer_borrowed, webrtc::PeerConnectionInterface::IceConnectionState);
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void (*onIceNetworkRouteChange)(void* observer_borrowed, webrtc::rffi::NetworkRoute);
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// Media events
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void (*onAddStream)(void* observer_borrowed, webrtc::MediaStreamInterface* stream_owned_rc);
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void (*onAddAudioRtpReceiver)(void* observer_borrowed, webrtc::MediaStreamTrackInterface* track_owned_rc);
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void (*onAddVideoRtpReceiver)(void* observer_borrowed, webrtc::MediaStreamTrackInterface* track_owned_rc);
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void (*onVideoFrame)(void* observer_borrowed, uint32_t track_id, RffiVideoFrameMetadata metadata, webrtc::VideoFrameBuffer* frame_buffer_borrowed);
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// RTP data events
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// Warning: this runs on the WebRTC network thread, so doing anything that
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// would block is dangerous, especially taking a lock that is also taken
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// while calling something that blocks on the network thread.
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void (*onRtpReceived)(void* observer_borrowed, uint8_t, uint16_t, uint32_t, uint32_t, const uint8_t* payload_borrowed, size_t);
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// Frame encryption
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size_t (*getMediaCiphertextBufferSize)(void* observer_borrowed, bool, size_t);
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bool (*encryptMedia)(void* observer_borrowed, bool, const uint8_t* plaintext_borrowed, size_t, uint8_t* ciphertext_out, size_t, size_t* ciphertext_size_out);
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size_t (*getMediaPlaintextBufferSize)(void* observer_borrowed, uint32_t, bool, size_t);
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bool (*decryptMedia)(void* observer_borrowed, uint32_t, bool, const uint8_t* ciphertext_borrowed, size_t, uint8_t* plaintext_out, size_t, size_t* plaintext_size_out);
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} PeerConnectionObserverCallbacks;
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// Passed-in observer must live at least as long as the PeerConnectionObserverRffi,
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// which is at least as long as the PeerConnection.
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RUSTEXPORT webrtc::rffi::PeerConnectionObserverRffi*
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Rust_createPeerConnectionObserver(void* observer_borrowed,
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const PeerConnectionObserverCallbacks* callbacks_borrowed,
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bool enable_frame_encryption,
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bool enable_video_frame_event,
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bool enable_video_frame_content);
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RUSTEXPORT void
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Rust_deletePeerConnectionObserver(webrtc::rffi::PeerConnectionObserverRffi* observer_owned);
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#endif /* RFFI_API_PEER_CONNECTION_OBSERVER_INTF_H__ */
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