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10542f21c8
Mechanically generated by running this command: tools_webrtc/do-renames.sh update all-renames.txt && git cl format Then manually updating: tools_webrtc/sanitizers/tsan_suppressions_webrtc.cc Bug: webrtc:10159 No-Presubmit: true No-Tree-Checks: true No-Try: true Change-Id: I54824cd91dada8fc3ee3d098f971bc319d477833 Reviewed-on: https://webrtc-review.googlesource.com/c/115653 Reviewed-by: Karl Wiberg <kwiberg@webrtc.org> Cr-Commit-Position: refs/heads/master@{#26226}
59 lines
2.7 KiB
C++
59 lines
2.7 KiB
C++
/*
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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_FRAME_DECRYPTOR_INTERFACE_H_
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#define API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_
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#include <vector>
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#include "api/array_view.h"
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#include "api/media_types.h"
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#include "rtc_base/ref_count.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. Additional data may be null.
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// Note: 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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~FrameDecryptorInterface() override {}
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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 GetMaxPlaintextSize. You may assume
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// that the frames are in order if SRTP is enabled. The stream is not provided
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// here and it is up to the implementor to transport this information to the
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// receiver if they care about it. You must set bytes_written to how many
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// bytes you wrote to in the frame buffer. 0 must be returned if successful
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// all other numbers can be selected by the implementer to represent error
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// codes.
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virtual int Decrypt(cricket::MediaType media_type,
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const std::vector<uint32_t>& csrcs,
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rtc::ArrayView<const uint8_t> additional_data,
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rtc::ArrayView<const uint8_t> encrypted_frame,
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rtc::ArrayView<uint8_t> frame,
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size_t* bytes_written) = 0;
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// Returns the total required length in bytes for the output of the
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// decryption. This can be larger than the actual number of bytes you need but
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// must never be smaller as it informs the size of the frame buffer.
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virtual size_t GetMaxPlaintextByteSize(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_FRAME_DECRYPTOR_INTERFACE_H_
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