/* * Copyright 2018 The WebRTC project authors. All Rights Reserved. * * Use of this source code is governed by a BSD-style license * that can be found in the LICENSE file in the root of the source * tree. An additional intellectual property rights grant can be found * in the file PATENTS. All contributing project authors may * be found in the AUTHORS file in the root of the source tree. */ #ifndef API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_ #define API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_ #include "api/array_view.h" #include "api/mediatypes.h" #include "rtc_base/refcount.h" namespace webrtc { // FrameEncryptorInterface allows users to provide a custom encryption // implementation to encrypt all outgoing audio and video frames. The user must // also provide a FrameDecryptorInterface to be able to decrypt the frames on // the receiving device. Note this is an additional layer of encryption in // addition to the standard SRTP mechanism and is not intended to be used // without it. Implementations of this interface will have the same lifetime as // the RTPSenders it is attached to. // This interface is not ready for production use. class FrameEncryptorInterface : public rtc::RefCountInterface { public: ~FrameEncryptorInterface() override {} // Attempts to encrypt the provided frame. You may assume the encrypted_frame // will match the size returned by GetOutputSize for a give frame. You may // assume that the frames will arrive in order if SRTP is enabled. The ssrc // will simply identify which stream the frame is travelling on. // TODO(benwright) integrate error codes. virtual bool Encrypt(cricket::MediaType media_type, uint32_t ssrc, rtc::ArrayView frame, rtc::ArrayView encrypted_frame) = 0; // Returns the total required length in bytes for the output of the // encryption. virtual size_t GetOutputSize(cricket::MediaType media_type, size_t frame_size) = 0; }; } // namespace webrtc #endif // API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_