| /* | 
 |  *  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. | 
 |  */ | 
 |  | 
 | #include "api/crypto/crypto_options.h" | 
 |  | 
 | #include "rtc_base/ssl_stream_adapter.h" | 
 |  | 
 | namespace webrtc { | 
 |  | 
 | CryptoOptions::CryptoOptions() {} | 
 |  | 
 | CryptoOptions::CryptoOptions(const CryptoOptions& other) { | 
 |   srtp = other.srtp; | 
 |   sframe = other.sframe; | 
 | } | 
 |  | 
 | CryptoOptions::~CryptoOptions() {} | 
 |  | 
 | // static | 
 | CryptoOptions CryptoOptions::NoGcm() { | 
 |   CryptoOptions options; | 
 |   options.srtp.enable_gcm_crypto_suites = false; | 
 |   return options; | 
 | } | 
 |  | 
 | std::vector<int> CryptoOptions::GetSupportedDtlsSrtpCryptoSuites() const { | 
 |   std::vector<int> crypto_suites; | 
 |   if (srtp.enable_gcm_crypto_suites) { | 
 |     crypto_suites.push_back(rtc::SRTP_AEAD_AES_256_GCM); | 
 |     crypto_suites.push_back(rtc::SRTP_AEAD_AES_128_GCM); | 
 |   } | 
 |   // Note: SRTP_AES128_CM_SHA1_80 is what is required to be supported (by | 
 |   // draft-ietf-rtcweb-security-arch), but SRTP_AES128_CM_SHA1_32 is allowed as | 
 |   // well, and saves a few bytes per packet if it ends up selected. | 
 |   // As the cipher suite is potentially insecure, it will only be used if | 
 |   // enabled by both peers. | 
 |   if (srtp.enable_aes128_sha1_32_crypto_cipher) { | 
 |     crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_32); | 
 |   } | 
 |   crypto_suites.push_back(rtc::SRTP_AES128_CM_SHA1_80); | 
 |   return crypto_suites; | 
 | } | 
 |  | 
 | bool CryptoOptions::operator==(const CryptoOptions& other) const { | 
 |   struct data_being_tested_for_equality { | 
 |     struct Srtp { | 
 |       bool enable_gcm_crypto_suites; | 
 |       bool enable_aes128_sha1_32_crypto_cipher; | 
 |       bool enable_encrypted_rtp_header_extensions; | 
 |     } srtp; | 
 |     struct SFrame { | 
 |       bool require_frame_encryption; | 
 |     } sframe; | 
 |   }; | 
 |   static_assert(sizeof(data_being_tested_for_equality) == sizeof(*this), | 
 |                 "Did you add something to CryptoOptions and forget to " | 
 |                 "update operator==?"); | 
 |  | 
 |   return srtp.enable_gcm_crypto_suites == other.srtp.enable_gcm_crypto_suites && | 
 |          srtp.enable_aes128_sha1_32_crypto_cipher == | 
 |              other.srtp.enable_aes128_sha1_32_crypto_cipher && | 
 |          srtp.enable_encrypted_rtp_header_extensions == | 
 |              other.srtp.enable_encrypted_rtp_header_extensions && | 
 |          sframe.require_frame_encryption == | 
 |              other.sframe.require_frame_encryption; | 
 | } | 
 |  | 
 | bool CryptoOptions::operator!=(const CryptoOptions& other) const { | 
 |   return !(*this == other); | 
 | } | 
 |  | 
 | }  // namespace webrtc |