Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector

talk/app/webrtc/webrtcsession_unittest.cc

 /*
  * libjingle
  * Copyright 2012 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
  *     this list of conditions and the following disclaimer in the documentation
  *     and/or other materials provided with the distribution.
  *  3. The name of the author may not be used to endorse or promote products
  *     derived from this software without specific prior written permission.
  *
  * THIS SOFTWARE IS PROVIDED BY THE AUTHOR ``AS IS'' AND ANY EXPRESS OR IMPLIED
  * WARRANTIES, INCLUDING, BUT NOT LIMITED TO, THE IMPLIED WARRANTIES OF
  * MERCHANTABILITY AND FITNESS FOR A PARTICULAR PURPOSE ARE DISCLAIMED. IN NO
  * EVENT SHALL THE AUTHOR BE LIABLE FOR ANY DIRECT, INDIRECT, INCIDENTAL,
  * SPECIAL, EXEMPLARY, OR CONSEQUENTIAL DAMAGES (INCLUDING, BUT NOT LIMITED TO,
  * PROCUREMENT OF SUBSTITUTE GOODS OR SERVICES; LOSS OF USE, DATA, OR PROFITS;
  * OR BUSINESS INTERRUPTION) HOWEVER CAUSED AND ON ANY THEORY OF LIABILITY,
  * WHETHER IN CONTRACT, STRICT LIABILITY, OR TORT (INCLUDING NEGLIGENCE OR
  * OTHERWISE) ARISING IN ANY WAY OUT OF THE USE OF THIS SOFTWARE, EVEN IF
  * ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
  */
 
+#include <utility>
 #include <vector>
 
 #include "talk/app/webrtc/audiotrack.h"
 #include "talk/app/webrtc/fakemediacontroller.h"
 #include "talk/app/webrtc/fakemetricsobserver.h"
 #include "talk/app/webrtc/jsepicecandidate.h"
 #include "talk/app/webrtc/jsepsessiondescription.h"
 #include "talk/app/webrtc/peerconnection.h"
 #include "talk/app/webrtc/sctputils.h"
 #include "talk/app/webrtc/streamcollection.h"
@@ skipping 380 matching lines @@
         allocator_.get(), &observer_));
     session_->SignalDataChannelOpenMessage.connect(
         this, &WebRtcSessionTest::OnDataChannelOpenMessage);
 
     EXPECT_EQ(PeerConnectionInterface::kIceConnectionNew,
         observer_.ice_connection_state_);
     EXPECT_EQ(PeerConnectionInterface::kIceGatheringNew,
         observer_.ice_gathering_state_);
 
     EXPECT_TRUE(session_->Initialize(options_, constraints_.get(),
-                                     dtls_identity_store.Pass(),
+                                     std::move(dtls_identity_store),
                                      rtc_configuration));
     session_->set_metrics_observer(metrics_observer_);
   }
 
   void OnDataChannelOpenMessage(const std::string& label,
                                 const InternalDataChannelInit& config) {
     last_data_channel_label_ = label;
     last_data_channel_config_ = config;
   }
 
@@ skipping 30 matching lines @@
     PeerConnectionInterface::RTCConfiguration configuration;
     if (cert_gen_method == ALREADY_GENERATED) {
       configuration.certificates.push_back(
           FakeDtlsIdentityStore::GenerateCertificate());
     } else if (cert_gen_method == DTLS_IDENTITY_STORE) {
       dtls_identity_store.reset(new FakeDtlsIdentityStore());
       dtls_identity_store->set_should_fail(false);
     } else {
       RTC_CHECK(false);
     }
-    Init(dtls_identity_store.Pass(), configuration);
+    Init(std::move(dtls_identity_store), configuration);
   }
 
   // Init with DTLS with a store that will fail to generate a certificate.
   void InitWithDtlsIdentityGenFail() {
     rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
         new FakeDtlsIdentityStore());
     dtls_identity_store->set_should_fail(true);
     PeerConnectionInterface::RTCConfiguration configuration;
-    Init(dtls_identity_store.Pass(), configuration);
+    Init(std::move(dtls_identity_store), configuration);
   }
 
   void InitWithDtmfCodec() {
     // Add kTelephoneEventCodec for dtmf test.
     const cricket::AudioCodec kTelephoneEventCodec(
         106, "telephone-event", 8000, 0, 1, 0);
     std::vector<cricket::AudioCodec> codecs;
     codecs.push_back(kTelephoneEventCodec);
     media_engine_->SetAudioCodecs(codecs);
     desc_factory_->set_audio_codecs(codecs);
@@ skipping 217 matching lines @@
                 video_content->protocol());
     }
   }
 
   // Set the internal fake description factories to do DTLS-SRTP.
   void SetFactoryDtlsSrtp() {
     desc_factory_->set_secure(cricket::SEC_DISABLED);
     std::string identity_name = "WebRTC" +
         rtc::ToString(rtc::CreateRandomId());
     // Confirmed to work with KT_RSA and KT_ECDSA.
-    tdesc_factory_->set_certificate(rtc::RTCCertificate::Create(
-        rtc::scoped_ptr<rtc::SSLIdentity>(rtc::SSLIdentity::Generate(
-            identity_name, rtc::KT_DEFAULT)).Pass()));
+    tdesc_factory_->set_certificate(
+        rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+            rtc::SSLIdentity::Generate(identity_name, rtc::KT_DEFAULT))));
     tdesc_factory_->set_secure(cricket::SEC_REQUIRED);
   }
 
   void VerifyFingerprintStatus(const cricket::SessionDescription* sdp,
                                bool expected) {
     const TransportInfo* audio = sdp->GetTransportInfoByName("audio");
     ASSERT_TRUE(audio != NULL);
     ASSERT_EQ(expected, audio->description.identity_fingerprint.get() != NULL);
     const TransportInfo* video = sdp->GetTransportInfoByName("video");
     ASSERT_TRUE(video != NULL);
@@ skipping 3518 matching lines @@
 }
 
 // TODO(bemasc): Add a TestIceStatesBundle with BUNDLE enabled.  That test
 // currently fails because upon disconnection and reconnection OnIceComplete is
 // called more than once without returning to IceGatheringGathering.
 
 INSTANTIATE_TEST_CASE_P(WebRtcSessionTests,
                         WebRtcSessionTest,
                         testing::Values(ALREADY_GENERATED,
                                         DTLS_IDENTITY_STORE));