Revert "Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32."

This reverts commit 6780c51b23516803dc27173d10ba98d018780447.

Reason for revert:

More details in crbug.com/810292

Original change's description:
> Use SRTP_AES128_CM_SHA1_80 by default instead of SRTP_AES128_CM_SHA1_32.
> 
> A field has been added to "CryptoOptions" to enable SRTP_AES128_CM_SHA1_32
> from native apps if really necessary.
> 
> R=​deadbeef@webrtc.org
> 
> Bug: webrtc:7670
> Change-Id: I36b6ab3e302fbf3cda2611ff196757e43a56e704
> Reviewed-on: https://webrtc-review.googlesource.com/41420
> Reviewed-by: Taylor Brandstetter <deadbeef@webrtc.org>
> Reviewed-by: Magnus Jedvert <magjed@webrtc.org>
> Commit-Queue: Joachim Bauch <jbauch@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#21952}

TBR=deadbeef@webrtc.org,magjed@webrtc.org,jbauch@webrtc.org

Change-Id: I643dbe023eca526f2cda4d97df045f2533741dd4
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7670
Reviewed-on: https://webrtc-review.googlesource.com/49880
Reviewed-by: Tommi <tommi@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#21961}
diff --git a/pc/peerconnection_integrationtest.cc b/pc/peerconnection_integrationtest.cc
index b090856..788b432 100644
--- a/pc/peerconnection_integrationtest.cc
+++ b/pc/peerconnection_integrationtest.cc
@@ -107,7 +107,7 @@
 
 // SRTP cipher name negotiated by the tests. This must be updated if the
 // default changes.
-static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_80;
+static const int kDefaultSrtpCryptoSuite = rtc::SRTP_AES128_CM_SHA1_32;
 static const int kDefaultSrtpCryptoSuiteGcm = rtc::SRTP_AEAD_AES_256_GCM;
 
 static const SocketAddress kDefaultLocalAddress("192.168.1.1", 0);
@@ -1152,10 +1152,13 @@
               expected_callee_received_video_frames);
   }
 
-  void TestNegotiatedCipherSuite(
-      const PeerConnectionFactory::Options& caller_options,
-      const PeerConnectionFactory::Options& callee_options,
-      int expected_cipher_suite) {
+  void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
+                                         bool remote_gcm_enabled,
+                                         int expected_cipher_suite) {
+    PeerConnectionFactory::Options caller_options;
+    caller_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled;
+    PeerConnectionFactory::Options callee_options;
+    callee_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled;
     ASSERT_TRUE(CreatePeerConnectionWrappersWithOptions(caller_options,
                                                         callee_options));
     rtc::scoped_refptr<webrtc::FakeMetricsObserver> caller_observer =
@@ -1174,17 +1177,6 @@
     caller()->pc()->RegisterUMAObserver(nullptr);
   }
 
-  void TestGcmNegotiationUsesCipherSuite(bool local_gcm_enabled,
-                                         bool remote_gcm_enabled,
-                                         int expected_cipher_suite) {
-    PeerConnectionFactory::Options caller_options;
-    caller_options.crypto_options.enable_gcm_crypto_suites = local_gcm_enabled;
-    PeerConnectionFactory::Options callee_options;
-    callee_options.crypto_options.enable_gcm_crypto_suites = remote_gcm_enabled;
-    TestNegotiatedCipherSuite(caller_options, callee_options,
-                              expected_cipher_suite);
-  }
-
  private:
   // |ss_| is used by |network_thread_| so it must be destroyed later.
   std::unique_ptr<rtc::VirtualSocketServer> ss_;
@@ -2318,38 +2310,6 @@
       kMaxWaitForFramesMs);
 }
 
-TEST_F(PeerConnectionIntegrationTest,
-       Aes128Sha1_32_CipherNotUsedWhenOnlyCallerSupported) {
-  ASSERT_NE(rtc::SRTP_AES128_CM_SHA1_32, kDefaultSrtpCryptoSuite);
-  PeerConnectionFactory::Options caller_options;
-  caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
-  PeerConnectionFactory::Options callee_options;
-  int expected_cipher_suite = kDefaultSrtpCryptoSuite;
-  TestNegotiatedCipherSuite(caller_options, callee_options,
-                            expected_cipher_suite);
-}
-
-TEST_F(PeerConnectionIntegrationTest,
-       Aes128Sha1_32_CipherNotUsedWhenOnlyCalleeSupported) {
-  ASSERT_NE(rtc::SRTP_AES128_CM_SHA1_32, kDefaultSrtpCryptoSuite);
-  PeerConnectionFactory::Options caller_options;
-  PeerConnectionFactory::Options callee_options;
-  callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
-  int expected_cipher_suite = kDefaultSrtpCryptoSuite;
-  TestNegotiatedCipherSuite(caller_options, callee_options,
-                            expected_cipher_suite);
-}
-
-TEST_F(PeerConnectionIntegrationTest, Aes128Sha1_32_CipherUsedWhenSupported) {
-  PeerConnectionFactory::Options caller_options;
-  caller_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
-  PeerConnectionFactory::Options callee_options;
-  callee_options.crypto_options.enable_aes128_sha1_32_crypto_cipher = true;
-  int expected_cipher_suite = rtc::SRTP_AES128_CM_SHA1_32;
-  TestNegotiatedCipherSuite(caller_options, callee_options,
-                            expected_cipher_suite);
-}
-
 // Test that a non-GCM cipher is used if both sides only support non-GCM.
 TEST_F(PeerConnectionIntegrationTest, NonGcmCipherUsedWhenGcmNotSupported) {
   bool local_gcm_enabled = false;