talk/app/webrtc/peerconnection_unittest.cc - Issue 1416673006: Convert internal representation of Srtp cryptos from string to int.

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: Add back an old function name to prevent build break in chromium. Created 5 years, 2 months 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 3cf66d64d8827b3c4e71887306844904409a3cdb..74ec2825f97f95e9b95cd5a9a51bd26068794871 100644

--- a/talk/app/webrtc/peerconnection_unittest.cc

+++ b/talk/app/webrtc/peerconnection_unittest.cc

@@ -111,7 +111,7 @@ 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 kDefaultSrtpCipher = rtc::SRTP_AES128_CM_SHA1_32;

pthatcher1 2015/11/11 19:59:40 Should we rename this to kDefaultSrtpCryptoSuite?

guoweis_webrtc 2015/11/17 01:21:15 Done.

#endif

static void RemoveLinesFromSdp(const std::string& line_start,

@@ -1282,7 +1282,7 @@ 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(),

@@ -1292,12 +1292,11 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12None) {

rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(

rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));

- EXPECT_EQ_WAIT(kDefaultSrtpCipher,

+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCipher),

initializing_client()->GetSrtpCipherStats(),

kMaxWaitForStatsMs);

EXPECT_EQ(1, init_observer->GetEnumCounter(

- webrtc::kEnumCounterAudioSrtpCipher,

- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));

+ webrtc::kEnumCounterAudioSrtpCipher, kDefaultSrtpCipher));

}

// Test that DTLS 1.2 is used if both ends support it.

@@ -1313,7 +1312,7 @@ 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(),

@@ -1323,12 +1322,11 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Both) {

rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(

rtc::SSL_PROTOCOL_DTLS_12, rtc::KT_DEFAULT)));

- EXPECT_EQ_WAIT(kDefaultSrtpCipher,

+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCipher),

initializing_client()->GetSrtpCipherStats(),

kMaxWaitForStatsMs);

EXPECT_EQ(1, init_observer->GetEnumCounter(

- webrtc::kEnumCounterAudioSrtpCipher,

- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));

+ webrtc::kEnumCounterAudioSrtpCipher, kDefaultSrtpCipher));

}

// Test that DTLS 1.0 is used if the initator supports DTLS 1.2 and the

@@ -1345,7 +1343,7 @@ 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(),

@@ -1355,12 +1353,11 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Init) {

rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(

rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));

- EXPECT_EQ_WAIT(kDefaultSrtpCipher,

+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCipher),

initializing_client()->GetSrtpCipherStats(),

kMaxWaitForStatsMs);

EXPECT_EQ(1, init_observer->GetEnumCounter(

- webrtc::kEnumCounterAudioSrtpCipher,

- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));

+ webrtc::kEnumCounterAudioSrtpCipher, kDefaultSrtpCipher));

}

// Test that DTLS 1.0 is used if the initator supports DTLS 1.0 and the

@@ -1377,7 +1374,7 @@ 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(),

@@ -1387,12 +1384,11 @@ TEST_F(JsepPeerConnectionP2PTestClient, GetDtls12Recv) {

rtc::SSLStreamAdapter::GetDefaultSslCipherForTest(

rtc::SSL_PROTOCOL_DTLS_10, rtc::KT_DEFAULT)));

- EXPECT_EQ_WAIT(kDefaultSrtpCipher,

+ EXPECT_EQ_WAIT(rtc::SrtpCryptoSuiteToName(kDefaultSrtpCipher),

initializing_client()->GetSrtpCipherStats(),

kMaxWaitForStatsMs);

EXPECT_EQ(1, init_observer->GetEnumCounter(

- webrtc::kEnumCounterAudioSrtpCipher,

- rtc::GetSrtpCryptoSuiteFromName(kDefaultSrtpCipher)));

+ webrtc::kEnumCounterAudioSrtpCipher, kDefaultSrtpCipher));

}

// This test sets up a call between two parties with audio, video and data.

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