This change integrates the FrameEncryptorInterface and the
FrameDecryptorInterface into the public WebRTC API surface. This change just addresses the headers and not the internal changes. Bug: webrtc:9681 Change-Id: I1db0172fe55ba378f62e7781c2b7dcdb93d63239 Reviewed-on: https://webrtc-review.googlesource.com/96622 Commit-Queue: Benjamin Wright <benwright@webrtc.org> Reviewed-by: Steve Anton <steveanton@webrtc.org> Cr-Commit-Position: refs/heads/master@{#24488}
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@ -52,6 +52,8 @@ rtc_static_library("libjingle_peerconnection_api") {
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"bitrate_constraints.h",
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"candidate.cc",
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"candidate.h",
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"crypto/framedecryptorinterface.h",
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"crypto/frameencryptorinterface.h",
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"cryptoparams.h",
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"datachannelinterface.cc",
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"datachannelinterface.h",
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@ -94,7 +96,6 @@ rtc_static_library("libjingle_peerconnection_api") {
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"umametrics.h",
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"videosourceproxy.h",
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]
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deps = [
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":array_view",
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":audio_options_api",
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52
api/crypto/framedecryptorinterface.h
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52
api/crypto/framedecryptorinterface.h
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@ -0,0 +1,52 @@
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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_FRAMEDECRYPTORINTERFACE_H_
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#define API_CRYPTO_FRAMEDECRYPTORINTERFACE_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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// FrameDecryptorInterface allows users to provide a custom decryption
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// implementation for all incoming audio and video frames. The user must also
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// provide a FrameEncryptorInterface to be able to encrypt the frames being
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// sent out of the device. Note this is an additional layer of encyrption in
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// addition to the standard SRTP mechanism and is not intended to be used
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// without it. You may assume that this interface will have the same lifetime
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// as the RTPReceiver it is attached to. It must only be attached to one
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// RTPReceiver.
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// Note:
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// This interface is not ready for production use.
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class FrameDecryptorInterface : public rtc::RefCountInterface {
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public:
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virtual ~FrameDecryptorInterface() {}
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// Attempts to decrypt the encrypted frame. You may assume the frame size will
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// be allocated to the size returned from GetOutputSize. You may assume that
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// the frames are in order if SRTP is enabled. The stream is not provided here
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// and it is up to the implementor to transport this information to the
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// receiver if they care about it.
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// TODO(benwright) integrate error codes
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virtual bool Decrypt(cricket::MediaType media_type,
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rtc::ArrayView<const uint8_t> encrypted_frame,
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rtc::ArrayView<uint8_t> frame) = 0;
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// Returns the total required length in bytes for the output of the
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// decryption.
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virtual size_t GetOutputSize(cricket::MediaType media_type,
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size_t encrypted_frame_size) = 0;
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};
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} // namespace webrtc
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#endif // API_CRYPTO_FRAMEDECRYPTORINTERFACE_H_
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50
api/crypto/frameencryptorinterface.h
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50
api/crypto/frameencryptorinterface.h
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@ -0,0 +1,50 @@
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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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