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Unified Diff: talk/app/webrtc/peerconnection_unittest.cc

Issue 1416673006: Convert internal representation of Srtp cryptos from string to int. (Closed) Base URL: https://chromium.googlesource.com/external/webrtc@master
Patch Set: change srtp crypto name conversion Created 5 years, 1 month ago
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Index: talk/app/webrtc/peerconnection_unittest.cc
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index 3193ffd898e71c1cdf23ccb3049c477c6ecd9811..4faf599907c50fd77bf6bfd45296293d70084646 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -113,7 +113,7 @@ static const char kDataChannelLabel[] = "data_channel";
#if !defined(THREAD_SANITIZER)
// SRTP cipher name negotiated by the tests. This must be updated if the
// default changes.
-static const char kDefaultSrtpCipher[] = "AES_CM_128_HMAC_SHA1_32";
+static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32;
#endif
static void RemoveLinesFromSdp(const std::string& line_start,
@@ -1327,7 +1327,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
initializing_client()->GetDtlsCipherStats(),
@@ -1337,12 +1337,12 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
- EXPECT_EQ_WAIT(kDefaultSrtpCipher,
+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
- EXPECT_EQ(1, init_observer->GetEnumCounter(
- webrtc::kEnumCounterAudioSrtpCipher,
- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+ EXPECT_EQ(1,
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
+ kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.2 is used if both ends support it.
@@ -1358,7 +1358,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)),
initializing_client()->GetDtlsCipherStats(),
@@ -1368,12 +1368,12 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)));
- EXPECT_EQ_WAIT(kDefaultSrtpCipher,
+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
- EXPECT_EQ(1, init_observer->GetEnumCounter(
- webrtc::kEnumCounterAudioSrtpCipher,
- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+ EXPECT_EQ(1,
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
+ kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the
@@ -1390,7 +1390,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
initializing_client()->GetDtlsCipherStats(),
@@ -1400,12 +1400,12 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
- EXPECT_EQ_WAIT(kDefaultSrtpCipher,
+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
- EXPECT_EQ(1, init_observer->GetEnumCounter(
- webrtc::kEnumCounterAudioSrtpCipher,
- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+ EXPECT_EQ(1,
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
+ kDefaultSrtpCryptoSuite));
}
// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the
@@ -1422,7 +1422,7 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
initializing_client()->pc()->RegisterUMAObserver(init_observer);
LocalP2PTest();
- EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::GetSslCipherSuiteName(
+ EXPECT_EQ_WAIT(rtc::SSLStreamAdapter::SslCipherSuiteToName(
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)),
initializing_client()->GetDtlsCipherStats(),
@@ -1432,12 +1432,12 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {
rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(
rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));
- EXPECT_EQ_WAIT(kDefaultSrtpCipher,
+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCryptoSuite),
initializing_client()->GetSrtpCipherStats(),
kMaxWaitForStatsMs);
- EXPECT_EQ(1, init_observer->GetEnumCounter(
- webrtc::kEnumCounterAudioSrtpCipher,
- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));
+ EXPECT_EQ(1,
+ init_observer->GetEnumCounter(webrtc::kEnumCounterAudioSrtpCipher,
+ kDefaultSrtpCryptoSuite));
}
// This test sets up a call between two parties with audio, video and data.
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