api/crypto/frame_decryptor_interface.h

/*
 *  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_FRAME_DECRYPTOR_INTERFACE_H_
#define API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_

#include <vector>

#include "api/array_view.h"
#include "api/media_types.h"
#include "rtc_base/ref_count.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:
  // The Status enum represents all possible states that can be
  // returned when attempting to decrypt a frame. kRecoverable indicates that
  // there was an error with the given frame and so it should not be passed to
  // the decoder, however it hints that the receive stream is still decryptable
  // which is important for determining when to send key frame requests.
  enum class Status { kOk, kRecoverable, kFailedToDecrypt };

  struct Result {
    Status status;
    size_t bytes_written;
  };

  ~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.
  // TODO(bugs.webrtc.org/10512) - Remove this after implementation rewrite.
  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) {
    bytes_written = 0;
    return 1;
  }

  // TODO(bugs.webrtc.org/10512) - Remove the other decrypt function and turn
  // this to a pure virtual.
  virtual Result 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;
    const int status = Decrypt(media_type, csrcs, additional_data,
                               encrypted_frame, frame, &bytes_written);
    Result decryption_result;
    decryption_result.bytes_written = bytes_written;
    if (status == 0) {
      decryption_result.status = Status::kOk;
    } else {
      decryption_result.status = Status::kFailedToDecrypt;
    }
    return decryption_result;
  }

  // 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_FRAME_DECRYPTOR_INTERFACE_H_