51 lines
2.1 KiB
C
51 lines
2.1 KiB
C
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/*
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* Copyright 2018 The WebRTC project authors. All Rights Reserved.
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*
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* Use of this source code is governed by a BSD-style license
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* that can be found in the LICENSE file in the root of the source
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* tree. An additional intellectual property rights grant can be found
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* in the file PATENTS. All contributing project authors may
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* be found in the AUTHORS file in the root of the source tree.
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*/
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#ifndef API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_
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#define API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_
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#include "api/array_view.h"
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#include "api/mediatypes.h"
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#include "rtc_base/refcount.h"
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namespace webrtc {
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// FrameEncryptorInterface allows users to provide a custom encryption
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// implementation to encrypt all outgoing audio and video frames. The user must
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// also provide a FrameDecryptorInterface to be able to decrypt the frames on
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// the receiving device. Note this is an additional layer of encryption in
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// addition to the standard SRTP mechanism and is not intended to be used
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// without it. Implementations of this interface will have the same lifetime as
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// the RTPSenders it is attached to.
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// This interface is not ready for production use.
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class FrameEncryptorInterface : public rtc::RefCountInterface {
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public:
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virtual ~FrameEncryptorInterface() {}
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// Attempts to encrypt the provided frame. You may assume the encrypted_frame
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// will match the size returned by GetOutputSize for a give frame. You may
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// assume that the frames will arrive in order if SRTP is enabled. The ssrc
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// will simply identify which stream the frame is travelling on.
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// TODO(benwright) integrate error codes.
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virtual bool Encrypt(cricket::MediaType media_type,
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uint32_t ssrc,
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rtc::ArrayView<const uint8_t> frame,
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rtc::ArrayView<uint8_t> encrypted_frame) = 0;
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// Returns the total required length in bytes for the output of the
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// encryption.
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virtual size_t GetOutputSize(cricket::MediaType media_type,
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size_t frame_size) = 0;
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};
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} // namespace webrtc
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#endif // API_CRYPTO_FRAMEENCRYPTORINTERFACE_H_
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