From 6bf0a5cb5034a7e684dcc3500e841785237ce2dd Mon Sep 17 00:00:00 2001
From: Daniel Baumann <daniel.baumann@progress-linux.org>
Date: Sun, 7 Apr 2024 19:32:43 +0200
Subject: Adding upstream version 1:115.7.0.

Signed-off-by: Daniel Baumann <daniel.baumann@progress-linux.org>
---
 third_party/libwebrtc/api/crypto/crypto_options.cc | 89 ++++++++++++++++++++++
 1 file changed, 89 insertions(+)
 create mode 100644 third_party/libwebrtc/api/crypto/crypto_options.cc

(limited to 'third_party/libwebrtc/api/crypto/crypto_options.cc')

diff --git a/third_party/libwebrtc/api/crypto/crypto_options.cc b/third_party/libwebrtc/api/crypto/crypto_options.cc
new file mode 100644
index 0000000000..22c5dd464b
--- /dev/null
+++ b/third_party/libwebrtc/api/crypto/crypto_options.cc
@@ -0,0 +1,89 @@
+/*
+ *  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;
+  // Note: kSrtpAes128CmSha1_80 is what is required to be supported (by
+  // draft-ietf-rtcweb-security-arch), but kSrtpAes128CmSha1_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::kSrtpAes128CmSha1_32);
+  }
+  if (srtp.enable_aes128_sha1_80_crypto_cipher) {
+    crypto_suites.push_back(rtc::kSrtpAes128CmSha1_80);
+  }
+
+  // Note: GCM cipher suites are not the top choice since they increase the
+  // packet size. In order to negotiate them the other side must not support
+  // kSrtpAes128CmSha1_80.
+  if (srtp.enable_gcm_crypto_suites) {
+    crypto_suites.push_back(rtc::kSrtpAeadAes256Gcm);
+    crypto_suites.push_back(rtc::kSrtpAeadAes128Gcm);
+  }
+  RTC_CHECK(!crypto_suites.empty());
+  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_aes128_sha1_80_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_aes128_sha1_80_crypto_cipher ==
+             other.srtp.enable_aes128_sha1_80_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
-- 
cgit v1.2.3