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-rw-r--r--third_party/libwebrtc/api/crypto/BUILD.gn49
-rw-r--r--third_party/libwebrtc/api/crypto/crypto_options.cc89
-rw-r--r--third_party/libwebrtc/api/crypto/crypto_options.h72
-rw-r--r--third_party/libwebrtc/api/crypto/frame_decryptor_interface.h76
-rw-r--r--third_party/libwebrtc/api/crypto/frame_decryptor_interface_gn/moz.build205
-rw-r--r--third_party/libwebrtc/api/crypto/frame_encryptor_interface.h54
-rw-r--r--third_party/libwebrtc/api/crypto/frame_encryptor_interface_gn/moz.build205
-rw-r--r--third_party/libwebrtc/api/crypto/options_gn/moz.build217
8 files changed, 967 insertions, 0 deletions
diff --git a/third_party/libwebrtc/api/crypto/BUILD.gn b/third_party/libwebrtc/api/crypto/BUILD.gn
new file mode 100644
index 0000000000..8d041ea059
--- /dev/null
+++ b/third_party/libwebrtc/api/crypto/BUILD.gn
@@ -0,0 +1,49 @@
+# Copyright (c) 2019 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.
+
+import("../../webrtc.gni")
+
+group("crypto") {
+ deps = [
+ ":frame_decryptor_interface",
+ ":frame_encryptor_interface",
+ ":options",
+ ]
+}
+
+rtc_library("options") {
+ visibility = [ "*" ]
+ sources = [
+ "crypto_options.cc",
+ "crypto_options.h",
+ ]
+ deps = [
+ "../../rtc_base:ssl",
+ "../../rtc_base/system:rtc_export",
+ ]
+}
+
+rtc_source_set("frame_decryptor_interface") {
+ visibility = [ "*" ]
+ sources = [ "frame_decryptor_interface.h" ]
+ deps = [
+ "..:array_view",
+ "..:rtp_parameters",
+ "../../rtc_base:refcount",
+ ]
+}
+
+rtc_source_set("frame_encryptor_interface") {
+ visibility = [ "*" ]
+ sources = [ "frame_encryptor_interface.h" ]
+ deps = [
+ "..:array_view",
+ "..:rtp_parameters",
+ "../../rtc_base:refcount",
+ ]
+}
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
diff --git a/third_party/libwebrtc/api/crypto/crypto_options.h b/third_party/libwebrtc/api/crypto/crypto_options.h
new file mode 100644
index 0000000000..317995134a
--- /dev/null
+++ b/third_party/libwebrtc/api/crypto/crypto_options.h
@@ -0,0 +1,72 @@
+/*
+ * 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_CRYPTO_OPTIONS_H_
+#define API_CRYPTO_CRYPTO_OPTIONS_H_
+
+#include <vector>
+
+#include "rtc_base/system/rtc_export.h"
+
+namespace webrtc {
+
+// CryptoOptions defines advanced cryptographic settings for native WebRTC.
+// These settings must be passed into PeerConnectionFactoryInterface::Options
+// and are only applicable to native use cases of WebRTC.
+struct RTC_EXPORT CryptoOptions {
+ CryptoOptions();
+ CryptoOptions(const CryptoOptions& other);
+ ~CryptoOptions();
+
+ // Helper method to return an instance of the CryptoOptions with GCM crypto
+ // suites disabled. This method should be used instead of depending on current
+ // default values set by the constructor.
+ static CryptoOptions NoGcm();
+
+ // Returns a list of the supported DTLS-SRTP Crypto suites based on this set
+ // of crypto options.
+ std::vector<int> GetSupportedDtlsSrtpCryptoSuites() const;
+
+ bool operator==(const CryptoOptions& other) const;
+ bool operator!=(const CryptoOptions& other) const;
+
+ // SRTP Related Peer Connection options.
+ struct Srtp {
+ // Enable GCM crypto suites from RFC 7714 for SRTP. GCM will only be used
+ // if both sides enable it.
+ bool enable_gcm_crypto_suites = false;
+
+ // If set to true, the (potentially insecure) crypto cipher
+ // kSrtpAes128CmSha1_32 will be included in the list of supported ciphers
+ // during negotiation. It will only be used if both peers support it and no
+ // other ciphers get preferred.
+ bool enable_aes128_sha1_32_crypto_cipher = false;
+
+ // The most commonly used cipher. Can be disabled, mostly for testing
+ // purposes.
+ bool enable_aes128_sha1_80_crypto_cipher = true;
+
+ // If set to true, encrypted RTP header extensions as defined in RFC 6904
+ // will be negotiated. They will only be used if both peers support them.
+ bool enable_encrypted_rtp_header_extensions = false;
+ } srtp;
+
+ // Options to be used when the FrameEncryptor / FrameDecryptor APIs are used.
+ struct SFrame {
+ // If set all RtpSenders must have an FrameEncryptor attached to them before
+ // they are allowed to send packets. All RtpReceivers must have a
+ // FrameDecryptor attached to them before they are able to receive packets.
+ bool require_frame_encryption = false;
+ } sframe;
+};
+
+} // namespace webrtc
+
+#endif // API_CRYPTO_CRYPTO_OPTIONS_H_
diff --git a/third_party/libwebrtc/api/crypto/frame_decryptor_interface.h b/third_party/libwebrtc/api/crypto/frame_decryptor_interface.h
new file mode 100644
index 0000000000..2f6bdac4b4
--- /dev/null
+++ b/third_party/libwebrtc/api/crypto/frame_decryptor_interface.h
@@ -0,0 +1,76 @@
+/*
+ * 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_FRAME_DECRYPTOR_INTERFACE_H_
+#define API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_
+
+#include <vector>
+
+#include "api/array_view.h"
+#include "api/media_types.h"
+#include "rtc_base/ref_count.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. Additional data may be null.
+class FrameDecryptorInterface : public rtc::RefCountInterface {
+ public:
+ // The Status enum represents all possible states that can be
+ // returned when attempting to decrypt a frame. kRecoverable indicates that
+ // there was an error with the given frame and so it should not be passed to
+ // the decoder, however it hints that the receive stream is still decryptable
+ // which is important for determining when to send key frame requests
+ // kUnknown should never be returned by the implementor.
+ enum class Status { kOk, kRecoverable, kFailedToDecrypt, kUnknown };
+
+ struct Result {
+ Result(Status status, size_t bytes_written)
+ : status(status), bytes_written(bytes_written) {}
+
+ bool IsOk() const { return status == Status::kOk; }
+
+ const Status status;
+ const size_t bytes_written;
+ };
+
+ ~FrameDecryptorInterface() override {}
+
+ // Attempts to decrypt the encrypted frame. You may assume the frame size will
+ // be allocated to the size returned from GetMaxPlaintextSize. 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. You must set bytes_written to how many
+ // bytes you wrote to in the frame buffer. kOk must be returned if successful,
+ // kRecoverable should be returned if the failure was due to something other
+ // than a decryption failure. kFailedToDecrypt should be returned in all other
+ // cases.
+ virtual Result Decrypt(cricket::MediaType media_type,
+ const std::vector<uint32_t>& csrcs,
+ rtc::ArrayView<const uint8_t> additional_data,
+ 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. This can be larger than the actual number of bytes you need but
+ // must never be smaller as it informs the size of the frame buffer.
+ virtual size_t GetMaxPlaintextByteSize(cricket::MediaType media_type,
+ size_t encrypted_frame_size) = 0;
+};
+
+} // namespace webrtc
+
+#endif // API_CRYPTO_FRAME_DECRYPTOR_INTERFACE_H_
diff --git a/third_party/libwebrtc/api/crypto/frame_decryptor_interface_gn/moz.build b/third_party/libwebrtc/api/crypto/frame_decryptor_interface_gn/moz.build
new file mode 100644
index 0000000000..1ce4cdd00a
--- /dev/null
+++ b/third_party/libwebrtc/api/crypto/frame_decryptor_interface_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+ ### This moz.build was AUTOMATICALLY GENERATED from a GN config, ###
+ ### DO NOT edit it by hand. ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+ "!/ipc/ipdl/_ipdlheaders",
+ "!/third_party/libwebrtc/gen",
+ "/ipc/chromium/src",
+ "/third_party/libwebrtc/",
+ "/third_party/libwebrtc/third_party/abseil-cpp/",
+ "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+ DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+ DEFINES["NDEBUG"] = True
+ DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+ DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+ DEFINES["ANDROID"] = True
+ DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+ DEFINES["HAVE_SYS_UIO_H"] = True
+ DEFINES["WEBRTC_ANDROID"] = True
+ DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+ DEFINES["WEBRTC_LINUX"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_GNU_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+ OS_LIBS += [
+ "log"
+ ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+ DEFINES["WEBRTC_MAC"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+ DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["USE_AURA"] = "1"
+ DEFINES["USE_GLIB"] = "1"
+ DEFINES["USE_NSS_CERTS"] = "1"
+ DEFINES["USE_OZONE"] = "1"
+ DEFINES["USE_UDEV"] = True
+ DEFINES["WEBRTC_LINUX"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_FILE_OFFSET_BITS"] = "64"
+ DEFINES["_LARGEFILE64_SOURCE"] = True
+ DEFINES["_LARGEFILE_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+ DEFINES["USE_GLIB"] = "1"
+ DEFINES["USE_OZONE"] = "1"
+ DEFINES["USE_X11"] = "1"
+ DEFINES["WEBRTC_BSD"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_FILE_OFFSET_BITS"] = "64"
+ DEFINES["_LARGEFILE64_SOURCE"] = True
+ DEFINES["_LARGEFILE_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+ DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+ DEFINES["NOMINMAX"] = True
+ DEFINES["NTDDI_VERSION"] = "0x0A000000"
+ DEFINES["PSAPI_VERSION"] = "2"
+ DEFINES["UNICODE"] = True
+ DEFINES["USE_AURA"] = "1"
+ DEFINES["WEBRTC_WIN"] = True
+ DEFINES["WIN32"] = True
+ DEFINES["WIN32_LEAN_AND_MEAN"] = True
+ DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+ DEFINES["WINVER"] = "0x0A00"
+ DEFINES["_ATL_NO_OPENGL"] = True
+ DEFINES["_CRT_RAND_S"] = True
+ DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+ DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+ DEFINES["_HAS_EXCEPTIONS"] = "0"
+ DEFINES["_HAS_NODISCARD"] = True
+ DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+ DEFINES["_SECURE_ATL"] = True
+ DEFINES["_UNICODE"] = True
+ DEFINES["_WIN32_WINNT"] = "0x0A00"
+ DEFINES["_WINDOWS"] = True
+ DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+ DEFINES["WEBRTC_ARCH_ARM64"] = True
+ DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+ DEFINES["WEBRTC_ARCH_ARM"] = True
+ DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+ DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+ DEFINES["MIPS32_LE"] = True
+ DEFINES["MIPS_FPU_LE"] = True
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+ DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+ DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+ DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+ OS_LIBS += [
+ "android_support",
+ "unwind"
+ ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+ OS_LIBS += [
+ "android_support"
+ ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+Library("frame_decryptor_interface_gn")
diff --git a/third_party/libwebrtc/api/crypto/frame_encryptor_interface.h b/third_party/libwebrtc/api/crypto/frame_encryptor_interface.h
new file mode 100644
index 0000000000..1452b80189
--- /dev/null
+++ b/third_party/libwebrtc/api/crypto/frame_encryptor_interface.h
@@ -0,0 +1,54 @@
+/*
+ * 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_FRAME_ENCRYPTOR_INTERFACE_H_
+#define API_CRYPTO_FRAME_ENCRYPTOR_INTERFACE_H_
+
+#include "api/array_view.h"
+#include "api/media_types.h"
+#include "rtc_base/ref_count.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. Additional data may be null.
+class FrameEncryptorInterface : public rtc::RefCountInterface {
+ public:
+ ~FrameEncryptorInterface() override {}
+
+ // Attempts to encrypt the provided frame. You may assume the encrypted_frame
+ // will match the size returned by GetMaxCiphertextByteSize 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. You
+ // must set bytes_written to the number of bytes you wrote in the
+ // encrypted_frame. 0 must be returned if successful all other numbers can be
+ // selected by the implementer to represent error codes.
+ virtual int Encrypt(cricket::MediaType media_type,
+ uint32_t ssrc,
+ rtc::ArrayView<const uint8_t> additional_data,
+ rtc::ArrayView<const uint8_t> frame,
+ rtc::ArrayView<uint8_t> encrypted_frame,
+ size_t* bytes_written) = 0;
+
+ // Returns the total required length in bytes for the output of the
+ // encryption. This can be larger than the actual number of bytes you need but
+ // must never be smaller as it informs the size of the encrypted_frame buffer.
+ virtual size_t GetMaxCiphertextByteSize(cricket::MediaType media_type,
+ size_t frame_size) = 0;
+};
+
+} // namespace webrtc
+
+#endif // API_CRYPTO_FRAME_ENCRYPTOR_INTERFACE_H_
diff --git a/third_party/libwebrtc/api/crypto/frame_encryptor_interface_gn/moz.build b/third_party/libwebrtc/api/crypto/frame_encryptor_interface_gn/moz.build
new file mode 100644
index 0000000000..b8385d1daa
--- /dev/null
+++ b/third_party/libwebrtc/api/crypto/frame_encryptor_interface_gn/moz.build
@@ -0,0 +1,205 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+ ### This moz.build was AUTOMATICALLY GENERATED from a GN config, ###
+ ### DO NOT edit it by hand. ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+ "!/ipc/ipdl/_ipdlheaders",
+ "!/third_party/libwebrtc/gen",
+ "/ipc/chromium/src",
+ "/third_party/libwebrtc/",
+ "/third_party/libwebrtc/third_party/abseil-cpp/",
+ "/tools/profiler/public"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+ DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+ DEFINES["NDEBUG"] = True
+ DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+ DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+ DEFINES["ANDROID"] = True
+ DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+ DEFINES["HAVE_SYS_UIO_H"] = True
+ DEFINES["WEBRTC_ANDROID"] = True
+ DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+ DEFINES["WEBRTC_LINUX"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_GNU_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+ OS_LIBS += [
+ "log"
+ ]
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+ DEFINES["WEBRTC_MAC"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+ DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["USE_AURA"] = "1"
+ DEFINES["USE_GLIB"] = "1"
+ DEFINES["USE_NSS_CERTS"] = "1"
+ DEFINES["USE_OZONE"] = "1"
+ DEFINES["USE_UDEV"] = True
+ DEFINES["WEBRTC_LINUX"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_FILE_OFFSET_BITS"] = "64"
+ DEFINES["_LARGEFILE64_SOURCE"] = True
+ DEFINES["_LARGEFILE_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+ DEFINES["USE_GLIB"] = "1"
+ DEFINES["USE_OZONE"] = "1"
+ DEFINES["USE_X11"] = "1"
+ DEFINES["WEBRTC_BSD"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_FILE_OFFSET_BITS"] = "64"
+ DEFINES["_LARGEFILE64_SOURCE"] = True
+ DEFINES["_LARGEFILE_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+ DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+ DEFINES["NOMINMAX"] = True
+ DEFINES["NTDDI_VERSION"] = "0x0A000000"
+ DEFINES["PSAPI_VERSION"] = "2"
+ DEFINES["UNICODE"] = True
+ DEFINES["USE_AURA"] = "1"
+ DEFINES["WEBRTC_WIN"] = True
+ DEFINES["WIN32"] = True
+ DEFINES["WIN32_LEAN_AND_MEAN"] = True
+ DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+ DEFINES["WINVER"] = "0x0A00"
+ DEFINES["_ATL_NO_OPENGL"] = True
+ DEFINES["_CRT_RAND_S"] = True
+ DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+ DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+ DEFINES["_HAS_EXCEPTIONS"] = "0"
+ DEFINES["_HAS_NODISCARD"] = True
+ DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+ DEFINES["_SECURE_ATL"] = True
+ DEFINES["_UNICODE"] = True
+ DEFINES["_WIN32_WINNT"] = "0x0A00"
+ DEFINES["_WINDOWS"] = True
+ DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+ DEFINES["WEBRTC_ARCH_ARM64"] = True
+ DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+ DEFINES["WEBRTC_ARCH_ARM"] = True
+ DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+ DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+ DEFINES["MIPS32_LE"] = True
+ DEFINES["MIPS_FPU_LE"] = True
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+ DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+ DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+ DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+ OS_LIBS += [
+ "android_support",
+ "unwind"
+ ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+ OS_LIBS += [
+ "android_support"
+ ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+Library("frame_encryptor_interface_gn")
diff --git a/third_party/libwebrtc/api/crypto/options_gn/moz.build b/third_party/libwebrtc/api/crypto/options_gn/moz.build
new file mode 100644
index 0000000000..aee1983c3e
--- /dev/null
+++ b/third_party/libwebrtc/api/crypto/options_gn/moz.build
@@ -0,0 +1,217 @@
+# This Source Code Form is subject to the terms of the Mozilla Public
+# License, v. 2.0. If a copy of the MPL was not distributed with this
+# file, You can obtain one at http://mozilla.org/MPL/2.0/.
+
+
+ ### This moz.build was AUTOMATICALLY GENERATED from a GN config, ###
+ ### DO NOT edit it by hand. ###
+
+COMPILE_FLAGS["OS_INCLUDES"] = []
+AllowCompilerWarnings()
+
+DEFINES["ABSL_ALLOCATOR_NOTHROW"] = "1"
+DEFINES["RTC_DAV1D_IN_INTERNAL_DECODER_FACTORY"] = True
+DEFINES["RTC_ENABLE_VP9"] = True
+DEFINES["WEBRTC_ENABLE_PROTOBUF"] = "0"
+DEFINES["WEBRTC_LIBRARY_IMPL"] = True
+DEFINES["WEBRTC_MOZILLA_BUILD"] = True
+DEFINES["WEBRTC_NON_STATIC_TRACE_EVENT_HANDLERS"] = "0"
+DEFINES["WEBRTC_STRICT_FIELD_TRIALS"] = "0"
+
+FINAL_LIBRARY = "webrtc"
+
+
+LOCAL_INCLUDES += [
+ "!/ipc/ipdl/_ipdlheaders",
+ "!/third_party/libwebrtc/gen",
+ "/ipc/chromium/src",
+ "/third_party/libwebrtc/",
+ "/third_party/libwebrtc/third_party/abseil-cpp/",
+ "/tools/profiler/public"
+]
+
+UNIFIED_SOURCES += [
+ "/third_party/libwebrtc/api/crypto/crypto_options.cc"
+]
+
+if not CONFIG["MOZ_DEBUG"]:
+
+ DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "0"
+ DEFINES["NDEBUG"] = True
+ DEFINES["NVALGRIND"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1":
+
+ DEFINES["DYNAMIC_ANNOTATIONS_ENABLED"] = "1"
+
+if CONFIG["OS_TARGET"] == "Android":
+
+ DEFINES["ANDROID"] = True
+ DEFINES["ANDROID_NDK_VERSION_ROLL"] = "r22_1"
+ DEFINES["HAVE_SYS_UIO_H"] = True
+ DEFINES["WEBRTC_ANDROID"] = True
+ DEFINES["WEBRTC_ANDROID_OPENSLES"] = True
+ DEFINES["WEBRTC_LINUX"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_GNU_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Darwin":
+
+ DEFINES["WEBRTC_MAC"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_LIBCPP_HAS_NO_ALIGNED_ALLOCATION"] = True
+ DEFINES["__ASSERT_MACROS_DEFINE_VERSIONS_WITHOUT_UNDERSCORES"] = "0"
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["USE_AURA"] = "1"
+ DEFINES["USE_GLIB"] = "1"
+ DEFINES["USE_NSS_CERTS"] = "1"
+ DEFINES["USE_OZONE"] = "1"
+ DEFINES["USE_UDEV"] = True
+ DEFINES["WEBRTC_LINUX"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_FILE_OFFSET_BITS"] = "64"
+ DEFINES["_LARGEFILE64_SOURCE"] = True
+ DEFINES["_LARGEFILE_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "OpenBSD":
+
+ DEFINES["USE_GLIB"] = "1"
+ DEFINES["USE_OZONE"] = "1"
+ DEFINES["USE_X11"] = "1"
+ DEFINES["WEBRTC_BSD"] = True
+ DEFINES["WEBRTC_POSIX"] = True
+ DEFINES["_FILE_OFFSET_BITS"] = "64"
+ DEFINES["_LARGEFILE64_SOURCE"] = True
+ DEFINES["_LARGEFILE_SOURCE"] = True
+ DEFINES["__STDC_CONSTANT_MACROS"] = True
+ DEFINES["__STDC_FORMAT_MACROS"] = True
+
+if CONFIG["OS_TARGET"] == "WINNT":
+
+ DEFINES["CERT_CHAIN_PARA_HAS_EXTRA_FIELDS"] = True
+ DEFINES["NOMINMAX"] = True
+ DEFINES["NTDDI_VERSION"] = "0x0A000000"
+ DEFINES["PSAPI_VERSION"] = "2"
+ DEFINES["UNICODE"] = True
+ DEFINES["USE_AURA"] = "1"
+ DEFINES["WEBRTC_WIN"] = True
+ DEFINES["WIN32"] = True
+ DEFINES["WIN32_LEAN_AND_MEAN"] = True
+ DEFINES["WINAPI_FAMILY"] = "WINAPI_FAMILY_DESKTOP_APP"
+ DEFINES["WINVER"] = "0x0A00"
+ DEFINES["_ATL_NO_OPENGL"] = True
+ DEFINES["_CRT_RAND_S"] = True
+ DEFINES["_CRT_SECURE_NO_DEPRECATE"] = True
+ DEFINES["_ENABLE_EXTENDED_ALIGNED_STORAGE"] = True
+ DEFINES["_HAS_EXCEPTIONS"] = "0"
+ DEFINES["_HAS_NODISCARD"] = True
+ DEFINES["_SCL_SECURE_NO_DEPRECATE"] = True
+ DEFINES["_SECURE_ATL"] = True
+ DEFINES["_UNICODE"] = True
+ DEFINES["_WIN32_WINNT"] = "0x0A00"
+ DEFINES["_WINDOWS"] = True
+ DEFINES["__STD_C"] = True
+
+if CONFIG["CPU_ARCH"] == "aarch64":
+
+ DEFINES["WEBRTC_ARCH_ARM64"] = True
+ DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "arm":
+
+ CXXFLAGS += [
+ "-mfpu=neon"
+ ]
+
+ DEFINES["WEBRTC_ARCH_ARM"] = True
+ DEFINES["WEBRTC_ARCH_ARM_V7"] = True
+ DEFINES["WEBRTC_HAS_NEON"] = True
+
+if CONFIG["CPU_ARCH"] == "mips32":
+
+ DEFINES["MIPS32_LE"] = True
+ DEFINES["MIPS_FPU_LE"] = True
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "mips64":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86":
+
+ DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64":
+
+ DEFINES["WEBRTC_ENABLE_AVX2"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Android":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Darwin":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "OpenBSD":
+
+ DEFINES["_DEBUG"] = True
+
+if CONFIG["MOZ_DEBUG"] == "1" and CONFIG["OS_TARGET"] == "WINNT":
+
+ DEFINES["_HAS_ITERATOR_DEBUGGING"] = "0"
+
+if CONFIG["MOZ_X11"] == "1" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["USE_X11"] = "1"
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Android":
+
+ OS_LIBS += [
+ "android_support",
+ "unwind"
+ ]
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Android":
+
+ CXXFLAGS += [
+ "-msse2"
+ ]
+
+ OS_LIBS += [
+ "android_support"
+ ]
+
+if CONFIG["CPU_ARCH"] == "aarch64" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "arm" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86" and CONFIG["OS_TARGET"] == "Linux":
+
+ CXXFLAGS += [
+ "-msse2"
+ ]
+
+ DEFINES["_GNU_SOURCE"] = True
+
+if CONFIG["CPU_ARCH"] == "x86_64" and CONFIG["OS_TARGET"] == "Linux":
+
+ DEFINES["_GNU_SOURCE"] = True
+
+Library("options_gn")