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}
This commit is contained in:
Benjamin Wright
2018-08-29 13:06:15 -07:00
committed by Commit Bot
parent ae3144c51f
commit ea0869145e
3 changed files with 104 additions and 1 deletions

View File

@ -52,6 +52,8 @@ rtc_static_library("libjingle_peerconnection_api") {
"bitrate_constraints.h",
"candidate.cc",
"candidate.h",
"crypto/framedecryptorinterface.h",
"crypto/frameencryptorinterface.h",
"cryptoparams.h",
"datachannelinterface.cc",
"datachannelinterface.h",
@ -94,7 +96,6 @@ rtc_static_library("libjingle_peerconnection_api") {
"umametrics.h",
"videosourceproxy.h",
]
deps = [
":array_view",
":audio_options_api",

View File

@ -0,0 +1,52 @@
/*
* 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 "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.
// Note:
// This interface is not ready for production use.
class FrameDecryptorInterface : public rtc::RefCountInterface {
public:
virtual ~FrameDecryptorInterface() {}
// Attempts to decrypt the encrypted frame. You may assume the frame size will
// be allocated to the size returned from GetOutputSize. 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.
// TODO(benwright) integrate error codes
virtual bool Decrypt(cricket::MediaType media_type,
rtc::ArrayView<const uint8_t> encrypted_frame,
rtc::ArrayView<uint8_t> frame) = 0;
// Returns the total required length in bytes for the output of the
// decryption.
virtual size_t GetOutputSize(cricket::MediaType media_type,
size_t encrypted_frame_size) = 0;
};
} // namespace webrtc
#endif // API_CRYPTO_FRAMEDECRYPTORINTERFACE_H_

View File

@ -0,0 +1,50 @@
/*
* 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:
virtual ~FrameEncryptorInterface() {}
// 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<const uint8_t> frame,
rtc::ArrayView<uint8_t> 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_