diff --git a/webrtc/BUILD.gn b/webrtc/BUILD.gn index e067791a18..e22ad24ea2 100644 --- a/webrtc/BUILD.gn +++ b/webrtc/BUILD.gn @@ -30,7 +30,6 @@ config("common_inherited_config") { defines = [ # TODO(kjellander): Cleanup unused ones and move defines closer to # the source when webrtc:4256 is completed. - "FEATURE_ENABLE_SSL", "FEATURE_ENABLE_VOICEMAIL", "EXPAT_RELATIVE_PATH", "GTEST_RELATIVE_PATH", @@ -132,12 +131,10 @@ config("common_config") { # targets, there's no point including the defines in that config here. # TODO(kjellander): Cleanup unused ones and move defines closer to the # source when webrtc:4256 is completed. - "HAVE_OPENSSL_SSL_H", "HAVE_SRTP", "HAVE_WEBRTC_VIDEO", "HAVE_WEBRTC_VOICE", "LOGGING_INSIDE_WEBRTC", - "SSL_USE_OPENSSL", "USE_WEBRTC_DEV_BRANCH", ] } else { diff --git a/webrtc/base/BUILD.gn b/webrtc/base/BUILD.gn index 16b010015d..248ef13f5f 100644 --- a/webrtc/base/BUILD.gn +++ b/webrtc/base/BUILD.gn @@ -35,21 +35,10 @@ config("rtc_base_approved_all_dependent_config") { } } -config("rtc_base_config") { - defines = [ "FEATURE_ENABLE_SSL" ] -} - config("rtc_base_chromium_config") { defines = [ "NO_MAIN_THREAD_WRAPPING" ] } -config("openssl_config") { - defines = [ - "SSL_USE_OPENSSL", - "HAVE_OPENSSL_SSL_H", - ] -} - config("rtc_base_all_dependent_config") { if (is_ios) { libs = [ @@ -370,16 +359,7 @@ rtc_static_library("rtc_base") { public_deps = [ ":rtc_base_approved", ] - - configs += [ - ":openssl_config", - ":rtc_base_config", - ] - - public_configs = [ - ":openssl_config", - ":rtc_base_config", - ] + public_configs = [] all_dependent_configs = [ ":rtc_base_all_dependent_config" ] @@ -537,7 +517,6 @@ rtc_static_library("rtc_base") { "proxyserver.h", "rollingaccumulator.h", "scopedptrcollection.h", - "sslconfig.h", "sslroots.h", "testbase64.h", "testclient.cc", diff --git a/webrtc/base/helpers.cc b/webrtc/base/helpers.cc index a8389d462e..aa6a6aea18 100644 --- a/webrtc/base/helpers.cc +++ b/webrtc/base/helpers.cc @@ -13,18 +13,7 @@ #include #include -#if defined(FEATURE_ENABLE_SSL) -#include "webrtc/base/sslconfig.h" -#if defined(SSL_USE_OPENSSL) #include -#else -#if defined(WEBRTC_WIN) -#define WIN32_LEAN_AND_MEAN -#include -#include -#endif // WEBRTC_WIN -#endif // else -#endif // FEATURE_ENABLED_SSL #include "webrtc/base/base64.h" #include "webrtc/base/basictypes.h" @@ -45,7 +34,6 @@ class RandomGenerator { virtual bool Generate(void* buf, size_t len) = 0; }; -#if defined(SSL_USE_OPENSSL) // The OpenSSL RNG. class SecureRandomGenerator : public RandomGenerator { public: @@ -57,79 +45,6 @@ class SecureRandomGenerator : public RandomGenerator { } }; -#else -#if defined(WEBRTC_WIN) -class SecureRandomGenerator : public RandomGenerator { - public: - SecureRandomGenerator() : advapi32_(NULL), rtl_gen_random_(NULL) {} - ~SecureRandomGenerator() { - FreeLibrary(advapi32_); - } - - virtual bool Init(const void* seed, size_t seed_len) { - // We don't do any additional seeding on Win32, we just use the CryptoAPI - // RNG (which is exposed as a hidden function off of ADVAPI32 so that we - // don't need to drag in all of CryptoAPI) - if (rtl_gen_random_) { - return true; - } - - advapi32_ = LoadLibrary(L"advapi32.dll"); - if (!advapi32_) { - return false; - } - - rtl_gen_random_ = reinterpret_cast( - GetProcAddress(advapi32_, "SystemFunction036")); - if (!rtl_gen_random_) { - FreeLibrary(advapi32_); - return false; - } - - return true; - } - virtual bool Generate(void* buf, size_t len) { - if (!rtl_gen_random_ && !Init(NULL, 0)) { - return false; - } - return (rtl_gen_random_(buf, static_cast(len)) != FALSE); - } - - private: - typedef BOOL (WINAPI *RtlGenRandomProc)(PVOID, ULONG); - HINSTANCE advapi32_; - RtlGenRandomProc rtl_gen_random_; -}; - -#elif !defined(FEATURE_ENABLE_SSL) - -// No SSL implementation -- use rand() -class SecureRandomGenerator : public RandomGenerator { - public: - virtual bool Init(const void* seed, size_t len) { - if (len >= 4) { - srand(*reinterpret_cast(seed)); - } else { - srand(*reinterpret_cast(seed)); - } - return true; - } - virtual bool Generate(void* buf, size_t len) { - char* bytes = reinterpret_cast(buf); - for (size_t i = 0; i < len; ++i) { - bytes[i] = static_cast(rand()); - } - return true; - } -}; - -#else - -#error No SSL implementation has been selected! - -#endif // WEBRTC_WIN -#endif - // A test random generator, for predictable output. class TestRandomGenerator : public RandomGenerator { public: diff --git a/webrtc/base/messagedigest.cc b/webrtc/base/messagedigest.cc index c08cab4ea9..5e8621c770 100644 --- a/webrtc/base/messagedigest.cc +++ b/webrtc/base/messagedigest.cc @@ -15,13 +15,7 @@ #include #include "webrtc/base/basictypes.h" -#include "webrtc/base/sslconfig.h" -#if SSL_USE_OPENSSL #include "webrtc/base/openssldigest.h" -#else -#include "webrtc/base/md5digest.h" -#include "webrtc/base/sha1digest.h" -#endif #include "webrtc/base/stringencode.h" namespace rtc { @@ -37,22 +31,12 @@ const char DIGEST_SHA_512[] = "sha-512"; static const size_t kBlockSize = 64; // valid for SHA-256 and down MessageDigest* MessageDigestFactory::Create(const std::string& alg) { -#if SSL_USE_OPENSSL MessageDigest* digest = new OpenSSLDigest(alg); if (digest->Size() == 0) { // invalid algorithm delete digest; digest = NULL; } return digest; -#else - MessageDigest* digest = NULL; - if (alg == DIGEST_MD5) { - digest = new Md5Digest(); - } else if (alg == DIGEST_SHA_1) { - digest = new Sha1Digest(); - } - return digest; -#endif } bool IsFips180DigestAlgorithm(const std::string& alg) { diff --git a/webrtc/base/openssladapter.cc b/webrtc/base/openssladapter.cc index 430191641a..d368186186 100644 --- a/webrtc/base/openssladapter.cc +++ b/webrtc/base/openssladapter.cc @@ -8,8 +8,6 @@ * be found in the AUTHORS file in the root of the source tree. */ -#if HAVE_OPENSSL_SSL_H - #include "webrtc/base/openssladapter.h" #if defined(WEBRTC_POSIX) @@ -965,5 +963,3 @@ OpenSSLAdapter::SetupSSLContext() { } } // namespace rtc - -#endif // HAVE_OPENSSL_SSL_H diff --git a/webrtc/base/openssldigest.cc b/webrtc/base/openssldigest.cc index 2618b7f9fd..0413f8f410 100644 --- a/webrtc/base/openssldigest.cc +++ b/webrtc/base/openssldigest.cc @@ -8,8 +8,6 @@ * be found in the AUTHORS file in the root of the source tree. */ -#if HAVE_OPENSSL_SSL_H - #include "webrtc/base/openssldigest.h" #include "webrtc/base/checks.h" @@ -118,5 +116,3 @@ bool OpenSSLDigest::GetDigestSize(const std::string& algorithm, } } // namespace rtc - -#endif // HAVE_OPENSSL_SSL_H diff --git a/webrtc/base/opensslidentity.cc b/webrtc/base/opensslidentity.cc index 2f1c565938..7b96f6a206 100644 --- a/webrtc/base/opensslidentity.cc +++ b/webrtc/base/opensslidentity.cc @@ -8,8 +8,6 @@ * be found in the AUTHORS file in the root of the source tree. */ -#if HAVE_OPENSSL_SSL_H - #include "webrtc/base/opensslidentity.h" #include @@ -576,5 +574,3 @@ bool OpenSSLIdentity::operator!=(const OpenSSLIdentity& other) const { } } // namespace rtc - -#endif // HAVE_OPENSSL_SSL_H diff --git a/webrtc/base/opensslstreamadapter.cc b/webrtc/base/opensslstreamadapter.cc index 3b3aa5d219..158315f150 100644 --- a/webrtc/base/opensslstreamadapter.cc +++ b/webrtc/base/opensslstreamadapter.cc @@ -8,8 +8,6 @@ * be found in the AUTHORS file in the root of the source tree. */ -#if HAVE_OPENSSL_SSL_H - #include "webrtc/base/opensslstreamadapter.h" #include @@ -45,11 +43,10 @@ namespace { namespace rtc { -#if (OPENSSL_VERSION_NUMBER >= 0x10001000L) -#define HAVE_DTLS_SRTP +#if (OPENSSL_VERSION_NUMBER < 0x10001000L) +#error "webrtc requires at least OpenSSL version 1.0.1, to support DTLS-SRTP" #endif -#ifdef HAVE_DTLS_SRTP // SRTP cipher suite table. |internal_name| is used to construct a // colon-separated profile strings which is needed by // SSL_CTX_set_tlsext_use_srtp(). @@ -65,7 +62,6 @@ static SrtpCipherMapEntry SrtpCipherMap[] = { {"SRTP_AEAD_AES_128_GCM", SRTP_AEAD_AES_128_GCM}, {"SRTP_AEAD_AES_256_GCM", SRTP_AEAD_AES_256_GCM}, {nullptr, 0}}; -#endif #ifdef OPENSSL_IS_BORINGSSL // Not used in production code. Actual time should be relative to Jan 1, 1970. @@ -432,7 +428,6 @@ bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label, bool use_context, uint8_t* result, size_t result_len) { -#ifdef HAVE_DTLS_SRTP int i; i = SSL_export_keying_material(ssl_, result, result_len, label.c_str(), @@ -443,14 +438,10 @@ bool OpenSSLStreamAdapter::ExportKeyingMaterial(const std::string& label, return false; return true; -#else - return false; -#endif } bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites( const std::vector& ciphers) { -#ifdef HAVE_DTLS_SRTP std::string internal_ciphers; if (state_ != SSL_NONE) @@ -481,13 +472,9 @@ bool OpenSSLStreamAdapter::SetDtlsSrtpCryptoSuites( srtp_ciphers_ = internal_ciphers; return true; -#else - return false; -#endif } bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) { -#ifdef HAVE_DTLS_SRTP RTC_DCHECK(state_ == SSL_CONNECTED); if (state_ != SSL_CONNECTED) return false; @@ -501,9 +488,6 @@ bool OpenSSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) { *crypto_suite = srtp_profile->id; RTC_DCHECK(!SrtpCryptoSuiteToName(*crypto_suite).empty()); return true; -#else - return false; -#endif } bool OpenSSLStreamAdapter::IsTlsConnected() { @@ -1096,14 +1080,12 @@ SSL_CTX* OpenSSLStreamAdapter::SetupSSLContext() { SSL_CTX_set_cipher_list(ctx, "DEFAULT:!NULL:!aNULL:!SHA256:!SHA384:!aECDH:!AESGCM+AES256:!aPSK"); -#ifdef HAVE_DTLS_SRTP if (!srtp_ciphers_.empty()) { if (SSL_CTX_set_tlsext_use_srtp(ctx, srtp_ciphers_.c_str())) { SSL_CTX_free(ctx); return NULL; } } -#endif return ctx; } @@ -1169,26 +1151,6 @@ int OpenSSLStreamAdapter::SSLVerifyCallback(int ok, X509_STORE_CTX* store) { return stream->VerifyPeerCertificate(); } -bool OpenSSLStreamAdapter::HaveDtls() { - return true; -} - -bool OpenSSLStreamAdapter::HaveDtlsSrtp() { -#ifdef HAVE_DTLS_SRTP - return true; -#else - return false; -#endif -} - -bool OpenSSLStreamAdapter::HaveExporter() { -#ifdef HAVE_DTLS_SRTP - return true; -#else - return false; -#endif -} - bool OpenSSLStreamAdapter::IsBoringSsl() { #ifdef OPENSSL_IS_BORINGSSL return true; @@ -1273,5 +1235,3 @@ void OpenSSLStreamAdapter::enable_time_callback_for_testing() { } } // namespace rtc - -#endif // HAVE_OPENSSL_SSL_H diff --git a/webrtc/base/opensslstreamadapter.h b/webrtc/base/opensslstreamadapter.h index e7d2174be8..d3edf3a670 100644 --- a/webrtc/base/opensslstreamadapter.h +++ b/webrtc/base/opensslstreamadapter.h @@ -109,10 +109,7 @@ class OpenSSLStreamAdapter : public SSLStreamAdapter { bool IsTlsConnected() override; - // Capabilities interfaces - static bool HaveDtls(); - static bool HaveDtlsSrtp(); - static bool HaveExporter(); + // Capabilities interfaces. static bool IsBoringSsl(); static bool IsAcceptableCipher(int cipher, KeyType key_type); diff --git a/webrtc/base/ssladapter.cc b/webrtc/base/ssladapter.cc index ba24e618ec..06fce54902 100644 --- a/webrtc/base/ssladapter.cc +++ b/webrtc/base/ssladapter.cc @@ -10,13 +10,7 @@ #include "webrtc/base/ssladapter.h" -#include "webrtc/base/sslconfig.h" - -#if SSL_USE_OPENSSL - -#include "openssladapter.h" - -#endif +#include "webrtc/base/openssladapter.h" /////////////////////////////////////////////////////////////////////////////// @@ -24,18 +18,11 @@ namespace rtc { SSLAdapter* SSLAdapter::Create(AsyncSocket* socket) { -#if SSL_USE_OPENSSL return new OpenSSLAdapter(socket); -#else // !SSL_USE_OPENSSL - delete socket; - return NULL; -#endif // SSL_USE_OPENSSL } /////////////////////////////////////////////////////////////////////////////// -#if SSL_USE_OPENSSL - bool InitializeSSL(VerificationCallback callback) { return OpenSSLAdapter::InitializeSSL(callback); } @@ -48,22 +35,6 @@ bool CleanupSSL() { return OpenSSLAdapter::CleanupSSL(); } -#else // !SSL_USE_OPENSSL - -bool InitializeSSL(VerificationCallback callback) { - return true; -} - -bool InitializeSSLThread() { - return true; -} - -bool CleanupSSL() { - return true; -} - -#endif // SSL_USE_OPENSSL - /////////////////////////////////////////////////////////////////////////////// } // namespace rtc diff --git a/webrtc/base/ssladapter_unittest.cc b/webrtc/base/ssladapter_unittest.cc index a6ec56ebc0..c591f19658 100644 --- a/webrtc/base/ssladapter_unittest.cc +++ b/webrtc/base/ssladapter_unittest.cc @@ -370,8 +370,6 @@ class SSLAdapterTestDTLS_ECDSA : public SSLAdapterTestBase { : SSLAdapterTestBase(rtc::SSL_MODE_DTLS, rtc::KeyParams::ECDSA()) {} }; -#if SSL_USE_OPENSSL - // Basic tests: TLS // Test that handshake works, using RSA @@ -419,5 +417,3 @@ TEST_F(SSLAdapterTestDTLS_ECDSA, TestDTLSTransfer) { TestHandshake(true); TestTransfer("Hello, world!"); } - -#endif // SSL_USE_OPENSSL diff --git a/webrtc/base/sslconfig.h b/webrtc/base/sslconfig.h deleted file mode 100644 index 6aabad07a8..0000000000 --- a/webrtc/base/sslconfig.h +++ /dev/null @@ -1,30 +0,0 @@ -/* - * Copyright 2012 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 WEBRTC_BASE_SSLCONFIG_H_ -#define WEBRTC_BASE_SSLCONFIG_H_ - -// If no preference has been indicated, default to SChannel on Windows and -// OpenSSL everywhere else, if it is available. -#if !defined(SSL_USE_SCHANNEL) && !defined(SSL_USE_OPENSSL) -#if defined(WEBRTC_WIN) - -#define SSL_USE_SCHANNEL 1 - -#else // defined(WEBRTC_WIN) - -#if defined(HAVE_OPENSSL_SSL_H) -#define SSL_USE_OPENSSL 1 -#endif - -#endif // !defined(WEBRTC_WIN) -#endif - -#endif // WEBRTC_BASE_SSLCONFIG_H_ diff --git a/webrtc/base/sslidentity.cc b/webrtc/base/sslidentity.cc index 645050a7e4..a5dd7b9ce6 100644 --- a/webrtc/base/sslidentity.cc +++ b/webrtc/base/sslidentity.cc @@ -17,14 +17,8 @@ #include "webrtc/base/base64.h" #include "webrtc/base/checks.h" #include "webrtc/base/logging.h" -#include "webrtc/base/sslconfig.h" -#include "webrtc/base/sslfingerprint.h" - -#if SSL_USE_OPENSSL - #include "webrtc/base/opensslidentity.h" - -#endif // SSL_USE_OPENSSL +#include "webrtc/base/sslfingerprint.h" namespace rtc { @@ -213,8 +207,6 @@ SSLCertChain::~SSLCertChain() { std::for_each(certs_.begin(), certs_.end(), DeleteCert); } -#if SSL_USE_OPENSSL - // static SSLCertificate* SSLCertificate::FromPEMString(const std::string& pem_string) { return OpenSSLCertificate::FromPEMString(pem_string); @@ -260,12 +252,6 @@ bool operator!=(const SSLIdentity& a, const SSLIdentity& b) { return !(a == b); } -#else // !SSL_USE_OPENSSL - -#error "No SSL implementation" - -#endif // SSL_USE_OPENSSL - // Read |n| bytes from ASN1 number string at *|pp| and return the numeric value. // Update *|pp| and *|np| to reflect number of read bytes. static inline int ASN1ReadInt(const unsigned char** pp, size_t* np, size_t n) { diff --git a/webrtc/base/sslstreamadapter.cc b/webrtc/base/sslstreamadapter.cc index c3ef3bc3ae..2f601c6257 100644 --- a/webrtc/base/sslstreamadapter.cc +++ b/webrtc/base/sslstreamadapter.cc @@ -9,14 +9,9 @@ */ #include "webrtc/base/sslstreamadapter.h" -#include "webrtc/base/sslconfig.h" - -#if SSL_USE_OPENSSL #include "webrtc/base/opensslstreamadapter.h" -#endif // SSL_USE_OPENSSL - /////////////////////////////////////////////////////////////////////////////// namespace rtc { @@ -101,11 +96,7 @@ CryptoOptions CryptoOptions::NoGcm() { } SSLStreamAdapter* SSLStreamAdapter::Create(StreamInterface* stream) { -#if SSL_USE_OPENSSL return new OpenSSLStreamAdapter(stream); -#else // !SSL_USE_OPENSSL - return NULL; -#endif // SSL_USE_OPENSSL } SSLStreamAdapter::SSLStreamAdapter(StreamInterface* stream) @@ -137,16 +128,6 @@ bool SSLStreamAdapter::GetDtlsSrtpCryptoSuite(int* crypto_suite) { return false; } -#if SSL_USE_OPENSSL -bool SSLStreamAdapter::HaveDtls() { - return OpenSSLStreamAdapter::HaveDtls(); -} -bool SSLStreamAdapter::HaveDtlsSrtp() { - return OpenSSLStreamAdapter::HaveDtlsSrtp(); -} -bool SSLStreamAdapter::HaveExporter() { - return OpenSSLStreamAdapter::HaveExporter(); -} bool SSLStreamAdapter::IsBoringSsl() { return OpenSSLStreamAdapter::IsBoringSsl(); } @@ -163,7 +144,6 @@ std::string SSLStreamAdapter::SslCipherSuiteToName(int cipher_suite) { void SSLStreamAdapter::enable_time_callback_for_testing() { OpenSSLStreamAdapter::enable_time_callback_for_testing(); } -#endif // SSL_USE_OPENSSL /////////////////////////////////////////////////////////////////////////////// diff --git a/webrtc/base/sslstreamadapter.h b/webrtc/base/sslstreamadapter.h index 391019165f..4f5ee02fe4 100644 --- a/webrtc/base/sslstreamadapter.h +++ b/webrtc/base/sslstreamadapter.h @@ -228,10 +228,9 @@ class SSLStreamAdapter : public StreamAdapterInterface { // SS_OPENING but IsTlsConnected should return true. virtual bool IsTlsConnected() = 0; - // Capabilities testing - static bool HaveDtls(); - static bool HaveDtlsSrtp(); - static bool HaveExporter(); + // Capabilities testing. + // Used to have "DTLS supported", "DTLS-SRTP supported" etc. methods, but now + // that's assumed. static bool IsBoringSsl(); // Returns true iff the supplied cipher is deemed to be strong. diff --git a/webrtc/base/sslstreamadapter_unittest.cc b/webrtc/base/sslstreamadapter_unittest.cc index 9d73abc304..82036913e0 100644 --- a/webrtc/base/sslstreamadapter_unittest.cc +++ b/webrtc/base/sslstreamadapter_unittest.cc @@ -19,7 +19,6 @@ #include "webrtc/base/gunit.h" #include "webrtc/base/helpers.h" #include "webrtc/base/ssladapter.h" -#include "webrtc/base/sslconfig.h" #include "webrtc/base/sslidentity.h" #include "webrtc/base/sslstreamadapter.h" #include "webrtc/base/stream.h" @@ -65,12 +64,6 @@ static const char kCERT_PEM[] = "UD0A8qfhfDM+LK6rPAnCsVN0NRDY3jvd6rzix9M=\n" "-----END CERTIFICATE-----\n"; -#define MAYBE_SKIP_TEST(feature) \ - if (!(rtc::SSLStreamAdapter::feature())) { \ - LOG(LS_INFO) << "Feature disabled... skipping"; \ - return; \ - } - class SSLStreamAdapterTestBase; class SSLDummyStreamBase : public rtc::StreamInterface, @@ -963,7 +956,6 @@ TEST_P(SSLStreamAdapterTestTLS, TestSetPeerCertificateDigestWithInvalidLength) { // Basic tests: DTLS // Test that we can make a handshake work TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnect) { - MAYBE_SKIP_TEST(HaveDtls); TestHandshake(); }; @@ -971,14 +963,12 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnect) { // each direction is lost. This gives us predictable loss // rather than having to tune random TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacket) { - MAYBE_SKIP_TEST(HaveDtls); SetLoseFirstPacket(true); TestHandshake(); }; // Test a handshake with loss and delay TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) { - MAYBE_SKIP_TEST(HaveDtls); SetLoseFirstPacket(true); SetDelay(2000); SetHandshakeWait(20000); @@ -988,7 +978,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSConnectWithLostFirstPacketDelay2s) { // Test a handshake with small MTU // Disabled due to https://code.google.com/p/webrtc/issues/detail?id=3910 TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) { - MAYBE_SKIP_TEST(HaveDtls); SetMtu(700); SetHandshakeWait(20000); TestHandshake(); @@ -996,20 +985,17 @@ TEST_P(SSLStreamAdapterTestDTLS, DISABLED_TestDTLSConnectWithSmallMtu) { // Test transfer -- trivial TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransfer) { - MAYBE_SKIP_TEST(HaveDtls); TestHandshake(); TestTransfer(100); }; TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithLoss) { - MAYBE_SKIP_TEST(HaveDtls); TestHandshake(); SetLoss(10); TestTransfer(100); }; TEST_P(SSLStreamAdapterTestDTLS, TestDTLSTransferWithDamage) { - MAYBE_SKIP_TEST(HaveDtls); SetDamage(); // Must be called first because first packet // write happens at end of handshake. TestHandshake(); @@ -1026,7 +1012,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSDelayedIdentityWithBogusDigest) { // Test DTLS-SRTP with all high ciphers TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); std::vector high; high.push_back(rtc::SRTP_AES128_CM_SHA1_80); SetDtlsSrtpCryptoSuites(high, true); @@ -1044,7 +1029,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHigh) { // Test DTLS-SRTP with all low ciphers TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); std::vector low; low.push_back(rtc::SRTP_AES128_CM_SHA1_32); SetDtlsSrtpCryptoSuites(low, true); @@ -1062,7 +1046,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpLow) { // Test DTLS-SRTP with a mismatch -- should not converge TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); std::vector high; high.push_back(rtc::SRTP_AES128_CM_SHA1_80); std::vector low; @@ -1079,7 +1062,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpHighLow) { // Test DTLS-SRTP with each side being mixed -- should select high TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); std::vector mixed; mixed.push_back(rtc::SRTP_AES128_CM_SHA1_80); mixed.push_back(rtc::SRTP_AES128_CM_SHA1_32); @@ -1098,7 +1080,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpMixed) { // Test DTLS-SRTP with all GCM-128 ciphers. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM128) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); std::vector gcm128; gcm128.push_back(rtc::SRTP_AEAD_AES_128_GCM); SetDtlsSrtpCryptoSuites(gcm128, true); @@ -1116,7 +1097,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM128) { // Test DTLS-SRTP with all GCM-256 ciphers. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM256) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); std::vector gcm256; gcm256.push_back(rtc::SRTP_AEAD_AES_256_GCM); SetDtlsSrtpCryptoSuites(gcm256, true); @@ -1134,7 +1114,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCM256) { // Test DTLS-SRTP with mixed GCM-128/-256 ciphers -- should not converge. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMismatch) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); std::vector gcm128; gcm128.push_back(rtc::SRTP_AEAD_AES_128_GCM); std::vector gcm256; @@ -1151,7 +1130,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMismatch) { // Test DTLS-SRTP with both GCM-128/-256 ciphers -- should select GCM-256. TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpGCMMixed) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); std::vector gcmBoth; gcmBoth.push_back(rtc::SRTP_AEAD_AES_256_GCM); gcmBoth.push_back(rtc::SRTP_AEAD_AES_128_GCM); @@ -1199,7 +1177,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSSrtpKeyAndSaltLengths) { // Test an exporter TEST_P(SSLStreamAdapterTestDTLS, TestDTLSExporter) { - MAYBE_SKIP_TEST(HaveExporter); TestHandshake(); unsigned char client_out[20]; unsigned char server_out[20]; @@ -1222,7 +1199,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestDTLSExporter) { // Test not yet valid certificates are not rejected. TEST_P(SSLStreamAdapterTestDTLS, TestCertNotYetValid) { - MAYBE_SKIP_TEST(HaveDtls); long one_day = 60 * 60 * 24; // Make the certificates not valid until one day later. ResetIdentitiesWithValidity(one_day, one_day); @@ -1231,7 +1207,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestCertNotYetValid) { // Test expired certificates are not rejected. TEST_P(SSLStreamAdapterTestDTLS, TestCertExpired) { - MAYBE_SKIP_TEST(HaveDtls); long one_day = 60 * 60 * 24; // Make the certificates already expired. ResetIdentitiesWithValidity(-one_day, -one_day); @@ -1240,15 +1215,12 @@ TEST_P(SSLStreamAdapterTestDTLS, TestCertExpired) { // Test data transfer using certs created from strings. TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestTransfer) { - MAYBE_SKIP_TEST(HaveDtls); TestHandshake(); TestTransfer(100); } // Test getting the remote certificate. TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) { - MAYBE_SKIP_TEST(HaveDtls); - // Peer certificates haven't been received yet. ASSERT_FALSE(GetPeerCertificate(true)); ASSERT_FALSE(GetPeerCertificate(false)); @@ -1282,7 +1254,6 @@ TEST_F(SSLStreamAdapterTestDTLSFromPEMStrings, TestDTLSGetPeerCertificate) { // Test getting the used DTLS ciphers. // DTLS 1.2 enabled for neither client nor server -> DTLS 1.0 will be used. TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuite) { - MAYBE_SKIP_TEST(HaveDtls); SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10); TestHandshake(); @@ -1302,7 +1273,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuite) { // Test getting the used DTLS 1.2 ciphers. // DTLS 1.2 enabled for client and server -> DTLS 1.2 will be used. TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Both) { - MAYBE_SKIP_TEST(HaveDtls); SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12); TestHandshake(); @@ -1321,7 +1291,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Both) { // DTLS 1.2 enabled for client only -> DTLS 1.0 will be used. TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Client) { - MAYBE_SKIP_TEST(HaveDtls); SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_12); TestHandshake(); @@ -1340,7 +1309,6 @@ TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Client) { // DTLS 1.2 enabled for server only -> DTLS 1.0 will be used. TEST_P(SSLStreamAdapterTestDTLS, TestGetSslCipherSuiteDtls12Server) { - MAYBE_SKIP_TEST(HaveDtls); SetupProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_10); TestHandshake(); diff --git a/webrtc/p2p/BUILD.gn b/webrtc/p2p/BUILD.gn index 7a25a75b74..e99440cf43 100644 --- a/webrtc/p2p/BUILD.gn +++ b/webrtc/p2p/BUILD.gn @@ -79,7 +79,7 @@ rtc_static_library("rtc_p2p") { "client/socketmonitor.h", ] - defines = [ "FEATURE_ENABLE_SSL" ] + defines = [] deps = [ "../base:rtc_base", diff --git a/webrtc/p2p/base/dtlstransportchannel_unittest.cc b/webrtc/p2p/base/dtlstransportchannel_unittest.cc index 2fc95d4e3b..7034343174 100644 --- a/webrtc/p2p/base/dtlstransportchannel_unittest.cc +++ b/webrtc/p2p/base/dtlstransportchannel_unittest.cc @@ -693,7 +693,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferSrtpTwoChannels) { // Connect with DTLS, and transfer some data. TEST_F(DtlsTransportChannelTest, TestTransferDtls) { - MAYBE_SKIP_TEST(HaveDtls); PrepareDtls(true, true, rtc::KT_DEFAULT); ASSERT_TRUE(Connect()); TestTransfer(0, 1000, 100, false); @@ -701,7 +700,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtls) { // Create two channels with DTLS, and transfer some data. TEST_F(DtlsTransportChannelTest, TestTransferDtlsTwoChannels) { - MAYBE_SKIP_TEST(HaveDtls); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); ASSERT_TRUE(Connect()); @@ -725,7 +723,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsNotOffered) { // Create two channels with DTLS 1.0 and check ciphers. TEST_F(DtlsTransportChannelTest, TestDtls12None) { - MAYBE_SKIP_TEST(HaveDtls); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_10); @@ -734,7 +731,6 @@ TEST_F(DtlsTransportChannelTest, TestDtls12None) { // Create two channels with DTLS 1.2 and check ciphers. TEST_F(DtlsTransportChannelTest, TestDtls12Both) { - MAYBE_SKIP_TEST(HaveDtls); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_12); @@ -743,7 +739,6 @@ TEST_F(DtlsTransportChannelTest, TestDtls12Both) { // Create two channels with DTLS 1.0 / DTLS 1.2 and check ciphers. TEST_F(DtlsTransportChannelTest, TestDtls12Client1) { - MAYBE_SKIP_TEST(HaveDtls); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_12, rtc::SSL_PROTOCOL_DTLS_10); @@ -752,7 +747,6 @@ TEST_F(DtlsTransportChannelTest, TestDtls12Client1) { // Create two channels with DTLS 1.2 / DTLS 1.0 and check ciphers. TEST_F(DtlsTransportChannelTest, TestDtls12Client2) { - MAYBE_SKIP_TEST(HaveDtls); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); SetMaxProtocolVersions(rtc::SSL_PROTOCOL_DTLS_10, rtc::SSL_PROTOCOL_DTLS_12); @@ -761,7 +755,6 @@ TEST_F(DtlsTransportChannelTest, TestDtls12Client2) { // Connect with DTLS, negotiate DTLS-SRTP, and transfer SRTP using bypass. TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); ASSERT_TRUE(Connect()); @@ -771,7 +764,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtp) { // Connect with DTLS-SRTP, transfer an invalid SRTP packet, and expects -1 // returned. TEST_F(DtlsTransportChannelTest, TestTransferDtlsInvalidSrtpPacket) { - MAYBE_SKIP_TEST(HaveDtls); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); ASSERT_TRUE(Connect()); @@ -781,7 +773,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsInvalidSrtpPacket) { // Connect with DTLS. A does DTLS-SRTP but B does not. TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpRejected) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, false); ASSERT_TRUE(Connect()); @@ -789,7 +780,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpRejected) { // Connect with DTLS. B does DTLS-SRTP but A does not. TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpNotOffered) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(false, true); ASSERT_TRUE(Connect()); @@ -797,7 +787,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpNotOffered) { // Create two channels with DTLS, negotiate DTLS-SRTP, and transfer bypass SRTP. TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpTwoChannels) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); @@ -808,7 +797,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpTwoChannels) { // Create a single channel with DTLS, and send normal data and SRTP data on it. TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpDemux) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); ASSERT_TRUE(Connect()); @@ -818,7 +806,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsSrtpDemux) { // Testing when the remote is passive. TEST_F(DtlsTransportChannelTest, TestTransferDtlsAnswererIsPassive) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); @@ -831,7 +818,6 @@ TEST_F(DtlsTransportChannelTest, TestTransferDtlsAnswererIsPassive) { // Testing with the legacy DTLS client which doesn't use setup attribute. // In this case legacy is the answerer. TEST_F(DtlsTransportChannelTest, TestDtlsSetupWithLegacyAsAnswerer) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); PrepareDtls(true, true, rtc::KT_DEFAULT); NegotiateWithLegacy(); rtc::SSLRole channel1_role; @@ -845,7 +831,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsSetupWithLegacyAsAnswerer) { // Testing re offer/answer after the session is estbalished. Roles will be // kept same as of the previous negotiation. TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromOfferer) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); @@ -862,7 +847,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromOfferer) { } TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromAnswerer) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); @@ -880,7 +864,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferFromAnswerer) { // Test that any change in role after the intial setup will result in failure. TEST_F(DtlsTransportChannelTest, TestDtlsRoleReversal) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); @@ -896,7 +879,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsRoleReversal) { // Test that using different setup attributes which results in similar ssl // role as the initial negotiation will result in success. TEST_F(DtlsTransportChannelTest, TestDtlsReOfferWithDifferentSetupAttr) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); @@ -912,7 +894,6 @@ TEST_F(DtlsTransportChannelTest, TestDtlsReOfferWithDifferentSetupAttr) { // Test that re-negotiation can be started before the clients become connected // in the first negotiation. TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); SetChannelCount(2); PrepareDtls(true, true, rtc::KT_DEFAULT); PrepareDtlsSrtp(true, true); @@ -932,7 +913,6 @@ TEST_F(DtlsTransportChannelTest, TestRenegotiateBeforeConnect) { // Test Certificates state after negotiation but before connection. TEST_F(DtlsTransportChannelTest, TestCertificatesBeforeConnect) { - MAYBE_SKIP_TEST(HaveDtls); PrepareDtls(true, true, rtc::KT_DEFAULT); Negotiate(); @@ -953,7 +933,6 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesBeforeConnect) { // Test Certificates state after connection. TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) { - MAYBE_SKIP_TEST(HaveDtls); PrepareDtls(true, true, rtc::KT_DEFAULT); ASSERT_TRUE(Connect()); @@ -984,7 +963,6 @@ TEST_F(DtlsTransportChannelTest, TestCertificatesAfterConnect) { // 60 seconds. The timer defaults to 1 second, but for WebRTC we should be // initializing it to 50ms. TEST_F(DtlsTransportChannelTest, TestRetransmissionSchedule) { - MAYBE_SKIP_TEST(HaveDtls); // We can only change the retransmission schedule with a recently-added // BoringSSL API. Skip the test if not built with BoringSSL. MAYBE_SKIP_TEST(IsBoringSsl); @@ -1025,7 +1003,6 @@ TEST_F(DtlsTransportChannelTest, TestRetransmissionSchedule) { // Test that a DTLS connection can be made even if the underlying transport // is connected before DTLS fingerprints/roles have been negotiated. TEST_F(DtlsTransportChannelTest, TestConnectBeforeNegotiate) { - MAYBE_SKIP_TEST(HaveDtls); PrepareDtls(true, true, rtc::KT_DEFAULT); ASSERT_TRUE(Connect(cricket::CONNECTIONROLE_ACTPASS, cricket::CONNECTIONROLE_ACTIVE, @@ -1158,7 +1135,6 @@ class DtlsEventOrderingTest }; TEST_P(DtlsEventOrderingTest, TestEventOrdering) { - MAYBE_SKIP_TEST(HaveDtls); TestEventOrdering(::testing::get<0>(GetParam()), ::testing::get<1>(GetParam())); } diff --git a/webrtc/pc/channel_unittest.cc b/webrtc/pc/channel_unittest.cc index 00ceb7d9d5..d6401a5935 100644 --- a/webrtc/pc/channel_unittest.cc +++ b/webrtc/pc/channel_unittest.cc @@ -25,12 +25,6 @@ #include "webrtc/p2p/base/faketransportcontroller.h" #include "webrtc/pc/channel.h" -#define MAYBE_SKIP_TEST(feature) \ - if (!(rtc::SSLStreamAdapter::feature())) { \ - LOG(LS_INFO) << "Feature disabled... skipping"; \ - return; \ - } - using cricket::CA_OFFER; using cricket::CA_PRANSWER; using cricket::CA_ANSWER; @@ -2243,32 +2237,26 @@ TEST_F(VoiceChannelSingleThreadTest, SendSrtcpMux) { } TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, 0); } TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, DTLS); } TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpGcmBoth) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS | GCM_CIPHER, DTLS | GCM_CIPHER); } TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpGcmOne) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS | GCM_CIPHER, DTLS); } TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpGcmTwo) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, DTLS | GCM_CIPHER); } TEST_F(VoiceChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX); } @@ -2576,32 +2564,26 @@ TEST_F(VoiceChannelDoubleThreadTest, SendSrtcpMux) { } TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, 0); } TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, DTLS); } TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpGcmBoth) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS | GCM_CIPHER, DTLS | GCM_CIPHER); } TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpGcmOne) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS | GCM_CIPHER, DTLS); } TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpGcmTwo) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, DTLS | GCM_CIPHER); } TEST_F(VoiceChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX); } @@ -2901,17 +2883,14 @@ TEST_F(VideoChannelSingleThreadTest, SendSrtpToRtp) { } TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, 0); } TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, DTLS); } TEST_F(VideoChannelSingleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX); } @@ -3133,17 +3112,14 @@ TEST_F(VideoChannelDoubleThreadTest, SendSrtpToRtp) { } TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, 0); } TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtp) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS, DTLS); } TEST_F(VideoChannelDoubleThreadTest, SendDtlsSrtpToDtlsSrtpRtcpMux) { - MAYBE_SKIP_TEST(HaveDtlsSrtp); Base::SendSrtpToSrtp(DTLS | RTCP_MUX, DTLS | RTCP_MUX); } diff --git a/webrtc/pc/peerconnection_unittest.cc b/webrtc/pc/peerconnection_unittest.cc index a172a605b5..19ef6e48db 100644 --- a/webrtc/pc/peerconnection_unittest.cc +++ b/webrtc/pc/peerconnection_unittest.cc @@ -46,12 +46,6 @@ #include "webrtc/pc/test/fakevideotrackrenderer.h" #include "webrtc/pc/test/mockpeerconnectionobservers.h" -#define MAYBE_SKIP_TEST(feature) \ - if (!(feature())) { \ - LOG(LS_INFO) << "Feature disabled... skipping"; \ - return; \ - } - using cricket::ContentInfo; using cricket::FakeWebRtcVideoDecoder; using cricket::FakeWebRtcVideoDecoderFactory; @@ -223,8 +217,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, rtc::Thread* network_thread, rtc::Thread* worker_thread) { std::unique_ptr cert_generator( - rtc::SSLStreamAdapter::HaveDtlsSrtp() ? - new FakeRTCCertificateGenerator() : nullptr); + new FakeRTCCertificateGenerator()); return CreateClientWithDtlsIdentityStore(id, constraints, options, config, std::move(cert_generator), true, @@ -237,8 +230,7 @@ class PeerConnectionTestClient : public webrtc::PeerConnectionObserver, rtc::Thread* network_thread, rtc::Thread* worker_thread) { std::unique_ptr cert_generator( - rtc::SSLStreamAdapter::HaveDtlsSrtp() ? - new FakeRTCCertificateGenerator() : nullptr); + new FakeRTCCertificateGenerator()); return CreateClientWithDtlsIdentityStore(id, nullptr, options, nullptr, std::move(cert_generator), false, @@ -1472,7 +1464,6 @@ class P2PTestConductor : public testing::Test { } void SetupAndVerifyDtlsCall() { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -1497,8 +1488,7 @@ class P2PTestConductor : public testing::Test { rtc_config.set_cpu_adaptation(false); std::unique_ptr cert_generator( - rtc::SSLStreamAdapter::HaveDtlsSrtp() ? - new FakeRTCCertificateGenerator() : nullptr); + new FakeRTCCertificateGenerator()); cert_generator->use_alternate_key(); // Make sure the new client is using a different certificate. @@ -1694,7 +1684,6 @@ TEST_F(P2PTestConductor, OneWayMediaCallWithoutConstraints) { // This test sets up a audio call initially and then upgrades to audio/video, // using DTLS. TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -1708,7 +1697,6 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsRenegotiate) { // This test sets up a call transfer to a new caller with a different DTLS // fingerprint. TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); SetupAndVerifyDtlsCall(); // Keeping the original peer around which will still send packets to the @@ -1727,7 +1715,6 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCallee) { // bundle is in effect in the restart, the channel can successfully reset its // DTLS-SRTP context. TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -1746,7 +1733,6 @@ TEST_F(P2PTestConductor, LocalP2PTestDtlsBundleInIceRestart) { // This test sets up a call transfer to a new callee with a different DTLS // fingerprint. TEST_F(P2PTestConductor, LocalP2PTestDtlsTransferCaller) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); SetupAndVerifyDtlsCall(); // Keeping the original peer around which will still send packets to the @@ -1780,7 +1766,6 @@ TEST_F(P2PTestConductor, LocalP2PTestReceiverDoesntSupportCVO) { // DTLS key agreement. The offerer don't support SDES. As a result, DTLS is // negotiated and used for transport. TEST_F(P2PTestConductor, LocalP2PTestOfferDtlsButNotSdes) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints setup_constraints; setup_constraints.AddMandatory(MediaConstraintsInterface::kEnableDtlsSrtp, true); @@ -2257,7 +2242,6 @@ TEST_F(P2PTestConductor, AddDataChannelAfterRenegotiation) { // negotiation is completed without error. #ifdef HAVE_SCTP TEST_F(P2PTestConductor, CreateOfferWithSctpDataChannel) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints constraints; constraints.SetMandatory( MediaConstraintsInterface::kEnableDtlsSrtp, true); diff --git a/webrtc/pc/peerconnectionendtoend_unittest.cc b/webrtc/pc/peerconnectionendtoend_unittest.cc index 4327e1de40..df41dac2a8 100644 --- a/webrtc/pc/peerconnectionendtoend_unittest.cc +++ b/webrtc/pc/peerconnectionendtoend_unittest.cc @@ -24,12 +24,6 @@ // Notice that mockpeerconnectionobservers.h must be included after the above! #include "webrtc/pc/test/mockpeerconnectionobservers.h" -#define MAYBE_SKIP_TEST(feature) \ - if (!(feature())) { \ - LOG(LS_INFO) << "Feature disabled... skipping"; \ - return; \ - } - using webrtc::DataChannelInterface; using webrtc::FakeConstraints; using webrtc::MediaConstraintsInterface; @@ -198,8 +192,6 @@ TEST_F(PeerConnectionEndToEndTest, CallWithLegacySdp) { // Verifies that a DataChannel created before the negotiation can transition to // "OPEN" and transfer data. TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - CreatePcs(); webrtc::DataChannelInit init; @@ -224,8 +216,6 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelBeforeNegotiate) { // Verifies that a DataChannel created after the negotiation can transition to // "OPEN" and transfer data. TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - CreatePcs(); webrtc::DataChannelInit init; @@ -257,8 +247,6 @@ TEST_F(PeerConnectionEndToEndTest, CreateDataChannelAfterNegotiate) { // Verifies that DataChannel IDs are even/odd based on the DTLS roles. TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - CreatePcs(); webrtc::DataChannelInit init; @@ -286,8 +274,6 @@ TEST_F(PeerConnectionEndToEndTest, DataChannelIdAssignment) { // there are multiple DataChannels. TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenTwoPairsOfDataChannels) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - CreatePcs(); webrtc::DataChannelInit init; @@ -409,8 +395,6 @@ TEST_F(PeerConnectionEndToEndTest, MessageTransferBetweenQuicDataChannels) { // See: https://bugs.chromium.org/p/webrtc/issues/detail?id=4453 TEST_F(PeerConnectionEndToEndTest, DISABLED_DataChannelFromOpenWorksAfterClose) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - CreatePcs(); webrtc::DataChannelInit init; @@ -437,8 +421,6 @@ TEST_F(PeerConnectionEndToEndTest, // reference count), no memory access violation will occur. // See: https://code.google.com/p/chromium/issues/detail?id=565048 TEST_F(PeerConnectionEndToEndTest, CloseDataChannelRemotelyWhileNotReferenced) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - CreatePcs(); webrtc::DataChannelInit init; diff --git a/webrtc/pc/peerconnectioninterface_unittest.cc b/webrtc/pc/peerconnectioninterface_unittest.cc index 09395a4b89..ba3529335f 100644 --- a/webrtc/pc/peerconnectioninterface_unittest.cc +++ b/webrtc/pc/peerconnectioninterface_unittest.cc @@ -318,12 +318,6 @@ static const char kDtlsSdesFallbackSdp[] = "inline:NzB4d1BINUAvLEw6UzF3WSJ+PSdFcGdUJShpX1Zj|2^20|1:32 " "dummy_session_params\r\n"; -#define MAYBE_SKIP_TEST(feature) \ - if (!(feature())) { \ - LOG(LS_INFO) << "Feature disabled... skipping"; \ - return; \ - } - using ::testing::Exactly; using cricket::StreamParams; using webrtc::AudioSourceInterface; @@ -2069,7 +2063,6 @@ TEST_F(PeerConnectionInterfaceTest, TestRejectDataChannelInAnswer) { // FireFox, use it as a remote session description, generate an answer and use // the answer as a local description. TEST_F(PeerConnectionInterfaceTest, ReceiveFireFoxOffer) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); FakeConstraints constraints; constraints.AddMandatory(webrtc::MediaConstraintsInterface::kEnableDtlsSrtp, true); diff --git a/webrtc/pc/test/peerconnectiontestwrapper.cc b/webrtc/pc/test/peerconnectiontestwrapper.cc index 6404e2b3fc..c433699652 100644 --- a/webrtc/pc/test/peerconnectiontestwrapper.cc +++ b/webrtc/pc/test/peerconnectiontestwrapper.cc @@ -76,8 +76,7 @@ bool PeerConnectionTestWrapper::CreatePc( } std::unique_ptr cert_generator( - rtc::SSLStreamAdapter::HaveDtlsSrtp() ? new FakeRTCCertificateGenerator() - : nullptr); + new FakeRTCCertificateGenerator()); peer_connection_ = peer_connection_factory_->CreatePeerConnection( config, constraints, std::move(port_allocator), std::move(cert_generator), this); diff --git a/webrtc/pc/webrtcsession_unittest.cc b/webrtc/pc/webrtcsession_unittest.cc index 053f64ceb3..efd5d0c870 100644 --- a/webrtc/pc/webrtcsession_unittest.cc +++ b/webrtc/pc/webrtcsession_unittest.cc @@ -50,12 +50,6 @@ #include "webrtc/pc/webrtcsession.h" #include "webrtc/pc/webrtcsessiondescriptionfactory.h" -#define MAYBE_SKIP_TEST(feature) \ - if (!(feature())) { \ - LOG(LS_INFO) << "Feature disabled... skipping"; \ - return; \ - } - using cricket::FakeVoiceMediaChannel; using cricket::TransportInfo; using rtc::SocketAddress; @@ -1850,7 +1844,6 @@ TEST_F(WebRtcSessionTest, TestSetRemoteNonSdesAnswerWhenSdesOn) { // Test that we accept an offer with a DTLS fingerprint when DTLS is on // and that we return an answer with a DTLS fingerprint. TEST_P(WebRtcSessionTest, TestReceiveDtlsOfferCreateDtlsAnswer) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); SendAudioVideoStream1(); InitWithDtls(GetParam()); SetFactoryDtlsSrtp(); @@ -1879,7 +1872,6 @@ TEST_P(WebRtcSessionTest, TestReceiveDtlsOfferCreateDtlsAnswer) { // Test that we set a local offer with a DTLS fingerprint when DTLS is on // and then we accept a remote answer with a DTLS fingerprint successfully. TEST_P(WebRtcSessionTest, TestCreateDtlsOfferReceiveDtlsAnswer) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); SendAudioVideoStream1(); InitWithDtls(GetParam()); SetFactoryDtlsSrtp(); @@ -1909,7 +1901,6 @@ TEST_P(WebRtcSessionTest, TestCreateDtlsOfferReceiveDtlsAnswer) { // Test that if we support DTLS and the other side didn't offer a fingerprint, // we will fail to set the remote description. TEST_P(WebRtcSessionTest, TestReceiveNonDtlsOfferWhenDtlsOn) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtls(GetParam()); cricket::MediaSessionOptions options; options.recv_video = true; @@ -1933,7 +1924,6 @@ TEST_P(WebRtcSessionTest, TestReceiveNonDtlsOfferWhenDtlsOn) { // Test that we return a failure when applying a local answer that doesn't have // a DTLS fingerprint when DTLS is required. TEST_P(WebRtcSessionTest, TestSetLocalNonDtlsAnswerWhenDtlsOn) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtls(GetParam()); SessionDescriptionInterface* offer = NULL; SessionDescriptionInterface* answer = NULL; @@ -1949,7 +1939,6 @@ TEST_P(WebRtcSessionTest, TestSetLocalNonDtlsAnswerWhenDtlsOn) { // Test that we return a failure when applying a remote answer that doesn't have // a DTLS fingerprint when DTLS is required. TEST_P(WebRtcSessionTest, TestSetRemoteNonDtlsAnswerWhenDtlsOn) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtls(GetParam()); SessionDescriptionInterface* offer = CreateOffer(); cricket::MediaSessionOptions options; @@ -3926,8 +3915,6 @@ TEST_F(WebRtcSessionTest, TestRtpDataChannel) { } TEST_P(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - configuration_.enable_rtp_data_channel = true; options_.disable_sctp_data_channels = false; @@ -3940,7 +3927,6 @@ TEST_P(WebRtcSessionTest, TestRtpDataChannelConstraintTakesPrecedence) { // Test that sctp_content_name/sctp_transport_name (used for stats) are correct // before and after BUNDLE is negotiated. TEST_P(WebRtcSessionTest, SctpContentAndTransportName) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); SetFactoryDtlsSrtp(); InitWithDtls(GetParam()); @@ -3974,8 +3960,6 @@ TEST_P(WebRtcSessionTest, SctpContentAndTransportName) { } TEST_P(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - InitWithDtls(GetParam()); std::unique_ptr offer(CreateOffer()); @@ -3984,7 +3968,6 @@ TEST_P(WebRtcSessionTest, TestCreateOfferWithSctpEnabledWithoutStreams) { } TEST_P(WebRtcSessionTest, TestCreateAnswerWithSctpInOfferAndNoStreams) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); SetFactoryDtlsSrtp(); InitWithDtls(GetParam()); @@ -4016,8 +3999,6 @@ TEST_P(WebRtcSessionTest, TestSctpDataChannelWithoutDtls) { // Test that if DTLS is enabled, we end up with an SctpTransport created // (and not an RtpDataChannel). TEST_P(WebRtcSessionTest, TestSctpDataChannelWithDtls) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - InitWithDtls(GetParam()); SetLocalDescriptionWithDataChannel(); @@ -4028,7 +4009,6 @@ TEST_P(WebRtcSessionTest, TestSctpDataChannelWithDtls) { // Test that if SCTP is disabled, we don't end up with an SctpTransport // created (or an RtpDataChannel). TEST_P(WebRtcSessionTest, TestDisableSctpDataChannels) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); options_.disable_sctp_data_channels = true; InitWithDtls(GetParam()); @@ -4038,7 +4018,6 @@ TEST_P(WebRtcSessionTest, TestDisableSctpDataChannels) { } TEST_P(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); const int new_send_port = 9998; const int new_recv_port = 7775; @@ -4080,8 +4059,6 @@ TEST_P(WebRtcSessionTest, TestSctpDataChannelSendPortParsing) { // WebRtcSession signals the SctpTransport creation request with the expected // config. TEST_P(WebRtcSessionTest, TestSctpDataChannelOpenMessage) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); - InitWithDtls(GetParam()); SetLocalDescriptionWithDataChannel(); @@ -4121,7 +4098,6 @@ TEST_P(WebRtcSessionTest, TestUsesProvidedCertificate) { // identity generation is finished (even if a certificate is provided this is // an async op). TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtls(GetParam()); EXPECT_TRUE(session_->waiting_for_certificate_for_testing()); @@ -4137,7 +4113,6 @@ TEST_P(WebRtcSessionTest, TestCreateOfferBeforeIdentityRequestReturnSuccess) { // identity generation is finished (even if a certificate is provided this is // an async op). TEST_P(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtls(GetParam()); SetFactoryDtlsSrtp(); @@ -4158,7 +4133,6 @@ TEST_P(WebRtcSessionTest, TestCreateAnswerBeforeIdentityRequestReturnSuccess) { // identity generation is finished (even if a certificate is provided this is // an async op). TEST_P(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtls(GetParam()); EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000); @@ -4170,7 +4144,6 @@ TEST_P(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnSuccess) { // Verifies that CreateOffer fails when CreateOffer is called after async // identity generation fails. TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtlsIdentityGenFail(); EXPECT_TRUE_WAIT(!session_->waiting_for_certificate_for_testing(), 1000); @@ -4183,7 +4156,6 @@ TEST_F(WebRtcSessionTest, TestCreateOfferAfterIdentityRequestReturnFailure) { // before async identity generation is finished. TEST_P(WebRtcSessionTest, TestMultipleCreateOfferBeforeIdentityRequestReturnSuccess) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); VerifyMultipleAsyncCreateDescription(GetParam(), CreateSessionDescriptionRequest::kOffer); } @@ -4192,7 +4164,6 @@ TEST_P(WebRtcSessionTest, // before async identity generation fails. TEST_F(WebRtcSessionTest, TestMultipleCreateOfferBeforeIdentityRequestReturnFailure) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); VerifyMultipleAsyncCreateDescriptionIdentityGenFailure( CreateSessionDescriptionRequest::kOffer); } @@ -4201,7 +4172,6 @@ TEST_F(WebRtcSessionTest, // before async identity generation is finished. TEST_P(WebRtcSessionTest, TestMultipleCreateAnswerBeforeIdentityRequestReturnSuccess) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); VerifyMultipleAsyncCreateDescription( GetParam(), CreateSessionDescriptionRequest::kAnswer); } @@ -4210,7 +4180,6 @@ TEST_P(WebRtcSessionTest, // before async identity generation fails. TEST_F(WebRtcSessionTest, TestMultipleCreateAnswerBeforeIdentityRequestReturnFailure) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); VerifyMultipleAsyncCreateDescriptionIdentityGenFailure( CreateSessionDescriptionRequest::kAnswer); } @@ -4254,7 +4223,6 @@ TEST_F(WebRtcSessionTest, TestCombinedAudioVideoBweConstraint) { // Tests that we can renegotiate new media content with ICE candidates in the // new remote SDP. TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtls(GetParam()); SetFactoryDtlsSrtp(); @@ -4284,7 +4252,6 @@ TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesInSdp) { // Tests that we can renegotiate new media content with ICE candidates separated // from the remote SDP. TEST_P(WebRtcSessionTest, TestRenegotiateNewMediaWithCandidatesSeparated) { - MAYBE_SKIP_TEST(rtc::SSLStreamAdapter::HaveDtlsSrtp); InitWithDtls(GetParam()); SetFactoryDtlsSrtp();