webrtc/api/crypto/framedecryptorinterface.h
Benjamin Wright 1f87ec6813 Add AEAD support to Frame Encryption API. Add Contribuitng Source To Decryptor.
This change allows supporting additional data for authentication and adds a
requirement for the contributing source to be provided during decryption.

Change-Id: Ifc19cb2d8a7d6c3715c83c95cf12f64df0bca454
Bug: webrtc:9681
Reviewed-on: https://webrtc-review.googlesource.com/100001
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Commit-Queue: Benjamin Wright <benwright@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#24712}
2018-09-12 21:09:30 +00:00

59 lines
2.7 KiB
C++

/*
* 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_FRAMEDECRYPTORINTERFACE_H_
#define API_CRYPTO_FRAMEDECRYPTORINTERFACE_H_
#include <vector>
#include "api/array_view.h"
#include "api/mediatypes.h"
#include "rtc_base/refcount.h"
namespace webrtc {
// FrameDecryptorInterface allows users to provide a custom decryption
// implementation for all incoming audio and video frames. The user must also
// provide a FrameEncryptorInterface to be able to encrypt the frames being
// sent out of the device. Note this is an additional layer of encyrption in
// addition to the standard SRTP mechanism and is not intended to be used
// without it. You may assume that this interface will have the same lifetime
// as the RTPReceiver it is attached to. It must only be attached to one
// RTPReceiver. Additional data may be null.
// Note: This interface is not ready for production use.
class FrameDecryptorInterface : public rtc::RefCountInterface {
public:
~FrameDecryptorInterface() override {}
// Attempts to decrypt the encrypted frame. You may assume the frame size will
// be allocated to the size returned from GetMaxPlaintextSize. You may assume
// that the frames are in order if SRTP is enabled. The stream is not provided
// here and it is up to the implementor to transport this information to the
// receiver if they care about it. You must set bytes_written to how many
// bytes you wrote to in the frame buffer. 0 must be returned if successful
// all other numbers can be selected by the implementer to represent error
// codes.
virtual int Decrypt(cricket::MediaType media_type,
const std::vector<uint32_t>& csrcs,
rtc::ArrayView<const uint8_t> additional_data,
rtc::ArrayView<const uint8_t> encrypted_frame,
rtc::ArrayView<uint8_t> frame,
size_t* bytes_written) = 0;
// Returns the total required length in bytes for the output of the
// decryption. This can be larger than the actual number of bytes you need but
// must never be smaller as it informs the size of the frame buffer.
virtual size_t GetMaxPlaintextByteSize(cricket::MediaType media_type,
size_t encrypted_frame_size) = 0;
};
} // namespace webrtc
#endif // API_CRYPTO_FRAMEDECRYPTORINTERFACE_H_