Don't call the Pass methods of rtc::Buffer, rtc::scoped_ptr, and rtc::ScopedVector
We can now use std::move instead!
This CL leaves the Pass methods in place; a follow-up CL will add deprecation annotations to them.
Review URL: https://codereview.webrtc.org/1460043002
Cr-Commit-Position: refs/heads/master@{#11064}
diff --git a/talk/app/webrtc/dtlsidentitystore.cc b/talk/app/webrtc/dtlsidentitystore.cc
index 2758779..390ec0d 100644
--- a/talk/app/webrtc/dtlsidentitystore.cc
+++ b/talk/app/webrtc/dtlsidentitystore.cc
@@ -27,6 +27,8 @@
#include "talk/app/webrtc/dtlsidentitystore.h"
+#include <utility>
+
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
#include "webrtc/base/logging.h"
@@ -72,7 +74,7 @@
// Posting to |this| avoids touching |store_| on threads other than
// |signaling_thread_| and thus avoids having to use locks.
IdentityResultMessageData* msg = new IdentityResultMessageData(
- new IdentityResult(key_type_, identity.Pass()));
+ new IdentityResult(key_type_, std::move(identity)));
signaling_thread_->Post(this, MSG_GENERATE_IDENTITY_RESULT, msg);
}
@@ -93,7 +95,7 @@
static_cast<IdentityResultMessageData*>(msg->pdata));
if (store_) {
store_->OnIdentityGenerated(pdata->data()->key_type_,
- pdata->data()->identity_.Pass());
+ std::move(pdata->data()->identity_));
}
}
break;
@@ -152,7 +154,7 @@
rtc::scoped_ptr<IdentityResultMessageData> pdata(
static_cast<IdentityResultMessageData*>(msg->pdata));
OnIdentityGenerated(pdata->data()->key_type_,
- pdata->data()->identity_.Pass());
+ std::move(pdata->data()->identity_));
break;
}
}
@@ -178,9 +180,9 @@
// Return identity async - post even though we are on |signaling_thread_|.
LOG(LS_VERBOSE) << "Using a free DTLS identity.";
++request_info_[key_type].gen_in_progress_counts_;
- IdentityResultMessageData* msg = new IdentityResultMessageData(
- new IdentityResult(key_type,
- request_info_[key_type].free_identity_.Pass()));
+ IdentityResultMessageData* msg =
+ new IdentityResultMessageData(new IdentityResult(
+ key_type, std::move(request_info_[key_type].free_identity_)));
signaling_thread_->Post(this, MSG_GENERATE_IDENTITY_RESULT, msg);
return;
}
@@ -228,7 +230,7 @@
// Return the result to the observer.
if (identity.get()) {
LOG(LS_VERBOSE) << "A DTLS identity is returned to an observer.";
- observer->OnSuccess(identity.Pass());
+ observer->OnSuccess(std::move(identity));
} else {
LOG(LS_WARNING) << "Failed to generate DTLS identity.";
observer->OnFailure(0);
diff --git a/talk/app/webrtc/dtlsidentitystore.h b/talk/app/webrtc/dtlsidentitystore.h
index a0eef98..2a5309d 100644
--- a/talk/app/webrtc/dtlsidentitystore.h
+++ b/talk/app/webrtc/dtlsidentitystore.h
@@ -30,6 +30,7 @@
#include <queue>
#include <string>
+#include <utility>
#include "webrtc/base/messagehandler.h"
#include "webrtc/base/messagequeue.h"
@@ -129,7 +130,7 @@
struct IdentityResult {
IdentityResult(rtc::KeyType key_type,
rtc::scoped_ptr<rtc::SSLIdentity> identity)
- : key_type_(key_type), identity_(identity.Pass()) {}
+ : key_type_(key_type), identity_(std::move(identity)) {}
rtc::KeyType key_type_;
rtc::scoped_ptr<rtc::SSLIdentity> identity_;
diff --git a/talk/app/webrtc/java/jni/peerconnection_jni.cc b/talk/app/webrtc/java/jni/peerconnection_jni.cc
index f444984..5b871d3 100644
--- a/talk/app/webrtc/java/jni/peerconnection_jni.cc
+++ b/talk/app/webrtc/java/jni/peerconnection_jni.cc
@@ -57,6 +57,7 @@
#define JNIEXPORT __attribute__((visibility("default")))
#include <limits>
+#include <utility>
#include "talk/app/webrtc/java/jni/classreferenceholder.h"
#include "talk/app/webrtc/java/jni/jni_helpers.h"
@@ -1631,7 +1632,7 @@
rtc::SSLIdentity::Generate(webrtc::kIdentityName, rtc::KT_ECDSA));
if (ssl_identity.get()) {
rtc_config.certificates.push_back(
- rtc::RTCCertificate::Create(ssl_identity.Pass()));
+ rtc::RTCCertificate::Create(std::move(ssl_identity)));
LOG(LS_INFO) << "ECDSA certificate created.";
} else {
// Failing to create certificate should not abort peer connection
diff --git a/talk/app/webrtc/peerconnection.cc b/talk/app/webrtc/peerconnection.cc
index 3a38248..85e03f9 100644
--- a/talk/app/webrtc/peerconnection.cc
+++ b/talk/app/webrtc/peerconnection.cc
@@ -28,8 +28,9 @@
#include "talk/app/webrtc/peerconnection.h"
#include <algorithm>
-#include <vector>
#include <cctype> // for isdigit
+#include <utility>
+#include <vector>
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/dtmfsender.h"
@@ -632,8 +633,8 @@
}
rtc::scoped_ptr<cricket::PortAllocator> allocator(
allocator_factory->CreatePortAllocator(stun_config, turn_config));
- return Initialize(configuration, constraints, allocator.Pass(),
- dtls_identity_store.Pass(), observer);
+ return Initialize(configuration, constraints, std::move(allocator),
+ std::move(dtls_identity_store), observer);
}
bool PeerConnection::Initialize(
@@ -649,7 +650,7 @@
}
observer_ = observer;
- port_allocator_ = allocator.Pass();
+ port_allocator_ = std::move(allocator);
std::vector<PortAllocatorFactoryInterface::StunConfiguration> stun_config;
std::vector<PortAllocatorFactoryInterface::TurnConfiguration> turn_config;
@@ -701,7 +702,7 @@
// Initialize the WebRtcSession. It creates transport channels etc.
if (!session_->Initialize(factory_->options(), constraints,
- dtls_identity_store.Pass(), configuration)) {
+ std::move(dtls_identity_store), configuration)) {
return false;
}
diff --git a/talk/app/webrtc/peerconnection_unittest.cc b/talk/app/webrtc/peerconnection_unittest.cc
index 55e4da8..49f4500 100644
--- a/talk/app/webrtc/peerconnection_unittest.cc
+++ b/talk/app/webrtc/peerconnection_unittest.cc
@@ -30,6 +30,7 @@
#include <algorithm>
#include <list>
#include <map>
+#include <utility>
#include <vector>
#include "talk/app/webrtc/dtmfsender.h"
@@ -151,7 +152,7 @@
const PeerConnectionFactory::Options* options,
rtc::scoped_ptr<webrtc::DtlsIdentityStoreInterface> dtls_identity_store) {
PeerConnectionTestClient* client(new PeerConnectionTestClient(id));
- if (!client->Init(constraints, options, dtls_identity_store.Pass())) {
+ if (!client->Init(constraints, options, std::move(dtls_identity_store))) {
delete client;
return nullptr;
}
@@ -167,7 +168,7 @@
: nullptr);
return CreateClientWithDtlsIdentityStore(id, constraints, options,
- dtls_identity_store.Pass());
+ std::move(dtls_identity_store));
}
~PeerConnectionTestClient() {
@@ -761,7 +762,7 @@
peer_connection_factory_->SetOptions(*options);
}
peer_connection_ = CreatePeerConnection(
- allocator_factory_.get(), constraints, dtls_identity_store.Pass());
+ allocator_factory_.get(), constraints, std::move(dtls_identity_store));
return peer_connection_.get() != nullptr;
}
@@ -776,7 +777,8 @@
ice_servers.push_back(ice_server);
return peer_connection_factory_->CreatePeerConnection(
- ice_servers, constraints, factory, dtls_identity_store.Pass(), this);
+ ice_servers, constraints, factory, std::move(dtls_identity_store),
+ this);
}
void HandleIncomingOffer(const std::string& msg) {
@@ -1129,7 +1131,8 @@
// Make sure the new client is using a different certificate.
return PeerConnectionTestClient::CreateClientWithDtlsIdentityStore(
- "New Peer: ", &setup_constraints, nullptr, dtls_identity_store.Pass());
+ "New Peer: ", &setup_constraints, nullptr,
+ std::move(dtls_identity_store));
}
void SendRtpData(webrtc::DataChannelInterface* dc, const std::string& data) {
diff --git a/talk/app/webrtc/peerconnectionfactory.cc b/talk/app/webrtc/peerconnectionfactory.cc
index d01f542..6d36c8b 100644
--- a/talk/app/webrtc/peerconnectionfactory.cc
+++ b/talk/app/webrtc/peerconnectionfactory.cc
@@ -27,6 +27,8 @@
#include "talk/app/webrtc/peerconnectionfactory.h"
+#include <utility>
+
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/localaudiosource.h"
#include "talk/app/webrtc/mediastream.h"
@@ -274,12 +276,8 @@
rtc::scoped_refptr<PeerConnection> pc(
new rtc::RefCountedObject<PeerConnection>(this));
- if (!pc->Initialize(
- configuration,
- constraints,
- chosen_allocator_factory,
- dtls_identity_store.Pass(),
- observer)) {
+ if (!pc->Initialize(configuration, constraints, chosen_allocator_factory,
+ std::move(dtls_identity_store), observer)) {
return NULL;
}
return PeerConnectionProxy::Create(signaling_thread(), pc);
diff --git a/talk/app/webrtc/peerconnectionfactory_unittest.cc b/talk/app/webrtc/peerconnectionfactory_unittest.cc
index f1d5353..d0018d9 100644
--- a/talk/app/webrtc/peerconnectionfactory_unittest.cc
+++ b/talk/app/webrtc/peerconnectionfactory_unittest.cc
@@ -26,6 +26,7 @@
*/
#include <string>
+#include <utility>
#include "talk/app/webrtc/fakeportallocatorfactory.h"
#include "talk/app/webrtc/mediastreaminterface.h"
@@ -158,9 +159,8 @@
rtc::scoped_ptr<FakeDtlsIdentityStore> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory->CreatePeerConnection(
- servers, nullptr, nullptr, dtls_identity_store.Pass(), &observer));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory->CreatePeerConnection(
+ servers, nullptr, nullptr, std::move(dtls_identity_store), &observer));
EXPECT_TRUE(pc.get() != nullptr);
}
@@ -180,11 +180,9 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, allocator_factory_.get(), std::move(dtls_identity_store),
+ &observer_));
EXPECT_TRUE(pc.get() != NULL);
StunConfigurations stun_configs;
webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
@@ -213,11 +211,9 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, allocator_factory_.get(), std::move(dtls_identity_store),
+ &observer_));
EXPECT_TRUE(pc.get() != NULL);
StunConfigurations stun_configs;
webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
@@ -251,11 +247,9 @@
ice_servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(ice_servers, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ ice_servers, nullptr, allocator_factory_.get(),
+ std::move(dtls_identity_store), &observer_));
EXPECT_TRUE(pc.get() != NULL);
StunConfigurations stun_configs;
webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
@@ -283,11 +277,9 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, allocator_factory_.get(), std::move(dtls_identity_store),
+ &observer_));
EXPECT_TRUE(pc.get() != NULL);
TurnConfigurations turn_configs;
webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn(
@@ -306,11 +298,9 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, allocator_factory_.get(), std::move(dtls_identity_store),
+ &observer_));
EXPECT_TRUE(pc.get() != NULL);
TurnConfigurations turn_configs;
webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn(
@@ -333,11 +323,9 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, allocator_factory_.get(), std::move(dtls_identity_store),
+ &observer_));
EXPECT_TRUE(pc.get() != NULL);
TurnConfigurations turn_configs;
webrtc::PortAllocatorFactoryInterface::TurnConfiguration turn1(
@@ -370,11 +358,9 @@
config.servers.push_back(ice_server);
rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store(
new FakeDtlsIdentityStore());
- rtc::scoped_refptr<PeerConnectionInterface> pc(
- factory_->CreatePeerConnection(config, nullptr,
- allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_));
+ rtc::scoped_refptr<PeerConnectionInterface> pc(factory_->CreatePeerConnection(
+ config, nullptr, allocator_factory_.get(), std::move(dtls_identity_store),
+ &observer_));
EXPECT_TRUE(pc.get() != NULL);
StunConfigurations stun_configs;
webrtc::PortAllocatorFactoryInterface::StunConfiguration stun1(
diff --git a/talk/app/webrtc/peerconnectionfactoryproxy.h b/talk/app/webrtc/peerconnectionfactoryproxy.h
index 9013ea5..db34ea7 100644
--- a/talk/app/webrtc/peerconnectionfactoryproxy.h
+++ b/talk/app/webrtc/peerconnectionfactoryproxy.h
@@ -29,6 +29,7 @@
#define TALK_APP_WEBRTC_PEERCONNECTIONFACTORYPROXY_H_
#include <string>
+#include <utility>
#include "talk/app/webrtc/peerconnectioninterface.h"
#include "talk/app/webrtc/proxy.h"
@@ -38,7 +39,7 @@
BEGIN_PROXY_MAP(PeerConnectionFactory)
PROXY_METHOD1(void, SetOptions, const Options&)
- // Can't use PROXY_METHOD5 because scoped_ptr must be Pass()ed.
+ // Can't use PROXY_METHOD5 because scoped_ptr must be moved.
// TODO(tommi,hbos): Use of templates to support scoped_ptr?
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection(
const PeerConnectionInterface::RTCConfiguration& a1,
@@ -84,7 +85,7 @@
DtlsIdentityStoreInterface* a4,
PeerConnectionObserver* a5) {
rtc::scoped_ptr<DtlsIdentityStoreInterface> ptr_a4(a4);
- return c_->CreatePeerConnection(a1, a2, a3, ptr_a4.Pass(), a5);
+ return c_->CreatePeerConnection(a1, a2, a3, std::move(ptr_a4), a5);
}
rtc::scoped_refptr<PeerConnectionInterface> CreatePeerConnection_ot2(
@@ -95,7 +96,8 @@
PeerConnectionObserver* a5) {
rtc::scoped_ptr<cricket::PortAllocator> ptr_a3(a3);
rtc::scoped_ptr<DtlsIdentityStoreInterface> ptr_a4(a4);
- return c_->CreatePeerConnection(a1, a2, ptr_a3.Pass(), ptr_a4.Pass(), a5);
+ return c_->CreatePeerConnection(a1, a2, std::move(ptr_a3),
+ std::move(ptr_a4), a5);
}
END_PROXY()
diff --git a/talk/app/webrtc/peerconnectioninterface.h b/talk/app/webrtc/peerconnectioninterface.h
index 100d8ce..799ca15 100644
--- a/talk/app/webrtc/peerconnectioninterface.h
+++ b/talk/app/webrtc/peerconnectioninterface.h
@@ -69,6 +69,7 @@
#define TALK_APP_WEBRTC_PEERCONNECTIONINTERFACE_H_
#include <string>
+#include <utility>
#include <vector>
#include "talk/app/webrtc/datachannelinterface.h"
@@ -604,7 +605,7 @@
PeerConnectionInterface::RTCConfiguration rtc_config;
rtc_config.servers = servers;
return CreatePeerConnection(rtc_config, constraints, allocator_factory,
- dtls_identity_store.Pass(), observer);
+ std::move(dtls_identity_store), observer);
}
virtual rtc::scoped_refptr<MediaStreamInterface>
diff --git a/talk/app/webrtc/peerconnectioninterface_unittest.cc b/talk/app/webrtc/peerconnectioninterface_unittest.cc
index 5e23931..930f538 100644
--- a/talk/app/webrtc/peerconnectioninterface_unittest.cc
+++ b/talk/app/webrtc/peerconnectioninterface_unittest.cc
@@ -26,6 +26,7 @@
*/
#include <string>
+#include <utility>
#include "talk/app/webrtc/audiotrack.h"
#include "talk/app/webrtc/fakeportallocatorfactory.h"
@@ -562,10 +563,9 @@
nullptr) && dtls) {
dtls_identity_store.reset(new FakeDtlsIdentityStore());
}
- pc_ = pc_factory_->CreatePeerConnection(servers, constraints,
- port_allocator_factory_.get(),
- dtls_identity_store.Pass(),
- &observer_);
+ pc_ = pc_factory_->CreatePeerConnection(
+ servers, constraints, port_allocator_factory_.get(),
+ std::move(dtls_identity_store), &observer_);
ASSERT_TRUE(pc_.get() != NULL);
observer_.SetPeerConnectionInterface(pc_.get());
EXPECT_EQ(PeerConnectionInterface::kStable, observer_.state_);
@@ -582,7 +582,7 @@
scoped_refptr<PeerConnectionInterface> pc;
pc = pc_factory_->CreatePeerConnection(
servers, nullptr, port_allocator_factory_.get(),
- dtls_identity_store.Pass(), &observer_);
+ std::move(dtls_identity_store), &observer_);
ASSERT_EQ(nullptr, pc);
}
diff --git a/talk/app/webrtc/statscollector_unittest.cc b/talk/app/webrtc/statscollector_unittest.cc
index 89e1c15..e7ee911 100644
--- a/talk/app/webrtc/statscollector_unittest.cc
+++ b/talk/app/webrtc/statscollector_unittest.cc
@@ -696,8 +696,7 @@
// Fake certificate to report
rtc::scoped_refptr<rtc::RTCCertificate> local_certificate(
rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::FakeSSLIdentity>(
- new rtc::FakeSSLIdentity(local_cert))
- .Pass()));
+ new rtc::FakeSSLIdentity(local_cert))));
// Configure MockWebRtcSession
EXPECT_CALL(session_,
diff --git a/talk/app/webrtc/test/fakedtlsidentitystore.h b/talk/app/webrtc/test/fakedtlsidentitystore.h
index 5e596ca..98074c7 100644
--- a/talk/app/webrtc/test/fakedtlsidentitystore.h
+++ b/talk/app/webrtc/test/fakedtlsidentitystore.h
@@ -29,6 +29,7 @@
#define TALK_APP_WEBRTC_TEST_FAKEDTLSIDENTITYSERVICE_H_
#include <string>
+#include <utility>
#include "talk/app/webrtc/dtlsidentitystore.h"
#include "talk/app/webrtc/peerconnectioninterface.h"
@@ -141,7 +142,7 @@
rtc::scoped_ptr<rtc::SSLIdentity> identity(
rtc::SSLIdentity::FromPEMStrings(pem_key, pem_cert));
- return rtc::RTCCertificate::Create(identity.Pass());
+ return rtc::RTCCertificate::Create(std::move(identity));
}
private:
diff --git a/talk/app/webrtc/test/peerconnectiontestwrapper.cc b/talk/app/webrtc/test/peerconnectiontestwrapper.cc
index 2eb24d9..032044c 100644
--- a/talk/app/webrtc/test/peerconnectiontestwrapper.cc
+++ b/talk/app/webrtc/test/peerconnectiontestwrapper.cc
@@ -25,6 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
+#include <utility>
+
#include "talk/app/webrtc/fakeportallocatorfactory.h"
#include "talk/app/webrtc/test/fakedtlsidentitystore.h"
#include "talk/app/webrtc/test/fakeperiodicvideocapturer.h"
@@ -97,7 +99,7 @@
new FakeDtlsIdentityStore() : nullptr);
peer_connection_ = peer_connection_factory_->CreatePeerConnection(
ice_servers, constraints, allocator_factory_.get(),
- dtls_identity_store.Pass(), this);
+ std::move(dtls_identity_store), this);
return peer_connection_.get() != NULL;
}
diff --git a/talk/app/webrtc/webrtcsession.cc b/talk/app/webrtc/webrtcsession.cc
index 5e946a2..29a4f33 100644
--- a/talk/app/webrtc/webrtcsession.cc
+++ b/talk/app/webrtc/webrtcsession.cc
@@ -724,7 +724,7 @@
// Use the |dtls_identity_store| to generate a certificate.
RTC_DCHECK(dtls_identity_store);
webrtc_session_desc_factory_.reset(new WebRtcSessionDescriptionFactory(
- signaling_thread(), channel_manager_, dtls_identity_store.Pass(),
+ signaling_thread(), channel_manager_, std::move(dtls_identity_store),
this, id()));
} else {
// Use the already generated certificate.
diff --git a/talk/app/webrtc/webrtcsession_unittest.cc b/talk/app/webrtc/webrtcsession_unittest.cc
index 6a38351..5997c53 100644
--- a/talk/app/webrtc/webrtcsession_unittest.cc
+++ b/talk/app/webrtc/webrtcsession_unittest.cc
@@ -25,6 +25,7 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
+#include <utility>
#include <vector>
#include "talk/app/webrtc/audiotrack.h"
@@ -425,7 +426,7 @@
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_);
}
@@ -476,7 +477,7 @@
} 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.
@@ -485,7 +486,7 @@
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() {
@@ -723,9 +724,9 @@
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);
}
diff --git a/talk/app/webrtc/webrtcsessiondescriptionfactory.cc b/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
index 25965af..bfdbb1a 100644
--- a/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
+++ b/talk/app/webrtc/webrtcsessiondescriptionfactory.cc
@@ -27,6 +27,8 @@
#include "talk/app/webrtc/webrtcsessiondescriptionfactory.h"
+#include <utility>
+
#include "talk/app/webrtc/dtlsidentitystore.h"
#include "talk/app/webrtc/jsep.h"
#include "talk/app/webrtc/jsepsessiondescription.h"
@@ -99,12 +101,12 @@
der_private_key.length());
rtc::scoped_ptr<rtc::SSLIdentity> identity(
rtc::SSLIdentity::FromPEMStrings(pem_key, pem_cert));
- SignalCertificateReady(rtc::RTCCertificate::Create(identity.Pass()));
+ SignalCertificateReady(rtc::RTCCertificate::Create(std::move(identity)));
}
void WebRtcIdentityRequestObserver::OnSuccess(
rtc::scoped_ptr<rtc::SSLIdentity> identity) {
- SignalCertificateReady(rtc::RTCCertificate::Create(identity.Pass()));
+ SignalCertificateReady(rtc::RTCCertificate::Create(std::move(identity)));
}
// static
@@ -143,7 +145,7 @@
// to just use a random number as session id and start version from
// |kInitSessionVersion|.
session_version_(kInitSessionVersion),
- dtls_identity_store_(dtls_identity_store.Pass()),
+ dtls_identity_store_(std::move(dtls_identity_store)),
identity_request_observer_(identity_request_observer),
session_(session),
session_id_(session_id),
@@ -177,7 +179,7 @@
: WebRtcSessionDescriptionFactory(
signaling_thread,
channel_manager,
- dtls_identity_store.Pass(),
+ std::move(dtls_identity_store),
new rtc::RefCountedObject<WebRtcIdentityRequestObserver>(),
session,
session_id,
diff --git a/talk/session/media/channel.cc b/talk/session/media/channel.cc
index 9180852..818a659 100644
--- a/talk/session/media/channel.cc
+++ b/talk/session/media/channel.cc
@@ -25,6 +25,8 @@
* ADVISED OF THE POSSIBILITY OF SUCH DAMAGE.
*/
+#include <utility>
+
#include "talk/session/media/channel.h"
#include "talk/media/base/constants.h"
@@ -555,7 +557,7 @@
// Avoid a copy by transferring the ownership of the packet data.
int message_id = (!rtcp) ? MSG_RTPPACKET : MSG_RTCPPACKET;
PacketMessageData* data = new PacketMessageData;
- data->packet = packet->Pass();
+ data->packet = std::move(*packet);
data->options = options;
worker_thread_->Post(this, message_id, data);
return true;
diff --git a/talk/session/media/channel_unittest.cc b/talk/session/media/channel_unittest.cc
index 35e7142..6b1d66f 100644
--- a/talk/session/media/channel_unittest.cc
+++ b/talk/session/media/channel_unittest.cc
@@ -175,17 +175,15 @@
if (flags1 & DTLS) {
// Confirmed to work with KT_RSA and KT_ECDSA.
- transport_controller1_.SetLocalCertificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT))
- .Pass()));
+ transport_controller1_.SetLocalCertificate(
+ rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+ rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT))));
}
if (flags2 & DTLS) {
// Confirmed to work with KT_RSA and KT_ECDSA.
- transport_controller2_.SetLocalCertificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT))
- .Pass()));
+ transport_controller2_.SetLocalCertificate(
+ rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+ rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT))));
}
// Add stream information (SSRC) to the local content but not to the remote
diff --git a/talk/session/media/mediasession_unittest.cc b/talk/session/media/mediasession_unittest.cc
index abb03e7..20b72e9 100644
--- a/talk/session/media/mediasession_unittest.cc
+++ b/talk/session/media/mediasession_unittest.cc
@@ -238,11 +238,9 @@
f2_.set_video_codecs(MAKE_VECTOR(kVideoCodecs2));
f2_.set_data_codecs(MAKE_VECTOR(kDataCodecs2));
tdf1_.set_certificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- new rtc::FakeSSLIdentity("id1")).Pass()));
+ rtc::scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id1"))));
tdf2_.set_certificate(rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- new rtc::FakeSSLIdentity("id2")).Pass()));
+ rtc::scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("id2"))));
}
// Create a video StreamParamsVec object with:
diff --git a/webrtc/base/buffer.cc b/webrtc/base/buffer.cc
index 90e687b..62855f1 100644
--- a/webrtc/base/buffer.cc
+++ b/webrtc/base/buffer.cc
@@ -11,6 +11,7 @@
#include "webrtc/base/buffer.h"
#include <cassert>
+#include <utility>
namespace rtc {
@@ -22,7 +23,9 @@
}
Buffer::Buffer(Buffer&& buf)
- : size_(buf.size()), capacity_(buf.capacity()), data_(buf.data_.Pass()) {
+ : size_(buf.size()),
+ capacity_(buf.capacity()),
+ data_(std::move(buf.data_)) {
assert(IsConsistent());
buf.OnMovedFrom();
}
diff --git a/webrtc/base/buffer.h b/webrtc/base/buffer.h
index 7b9402b..076fa08 100644
--- a/webrtc/base/buffer.h
+++ b/webrtc/base/buffer.h
@@ -104,7 +104,7 @@
assert(buf.IsConsistent());
size_ = buf.size_;
capacity_ = buf.capacity_;
- data_ = buf.data_.Pass();
+ data_ = std::move(buf.data_);
buf.OnMovedFrom();
return *this;
}
@@ -164,12 +164,12 @@
return;
scoped_ptr<uint8_t[]> new_data(new uint8_t[capacity]);
std::memcpy(new_data.get(), data_.get(), size_);
- data_ = new_data.Pass();
+ data_ = std::move(new_data);
capacity_ = capacity;
assert(IsConsistent());
}
- // We can't call std::move(b), so call b.Pass() instead to do the same job.
+ // b.Pass() does the same thing as std::move(b).
Buffer&& Pass() {
assert(IsConsistent());
return static_cast<Buffer&&>(*this);
diff --git a/webrtc/base/messagehandler.h b/webrtc/base/messagehandler.h
index df82b4e..b55b229 100644
--- a/webrtc/base/messagehandler.h
+++ b/webrtc/base/messagehandler.h
@@ -11,6 +11,8 @@
#ifndef WEBRTC_BASE_MESSAGEHANDLER_H_
#define WEBRTC_BASE_MESSAGEHANDLER_H_
+#include <utility>
+
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/scoped_ptr.h"
@@ -54,8 +56,8 @@
: public MessageHandler {
public:
explicit FunctorMessageHandler(const FunctorT& functor) : functor_(functor) {}
- virtual void OnMessage(Message* msg) { result_ = functor_().Pass(); }
- rtc::scoped_ptr<ReturnT> result() { return result_.Pass(); }
+ virtual void OnMessage(Message* msg) { result_ = std::move(functor_()); }
+ rtc::scoped_ptr<ReturnT> result() { return std::move(result_); }
private:
FunctorT functor_;
diff --git a/webrtc/base/rtccertificate_unittests.cc b/webrtc/base/rtccertificate_unittests.cc
index 3e9439f..84c8544 100644
--- a/webrtc/base/rtccertificate_unittests.cc
+++ b/webrtc/base/rtccertificate_unittests.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <utility>
+
#include "webrtc/base/checks.h"
#include "webrtc/base/fakesslidentity.h"
#include "webrtc/base/gunit.h"
@@ -76,7 +78,7 @@
params.key_params = KeyParams::ECDSA();
scoped_ptr<SSLIdentity> identity(SSLIdentity::GenerateForTest(params));
- return RTCCertificate::Create(identity.Pass());
+ return RTCCertificate::Create(std::move(identity));
}
};
@@ -86,7 +88,7 @@
scoped_ptr<SSLIdentity> identity(
SSLIdentity::Generate(kTestCertCommonName, KeyParams::ECDSA()));
scoped_refptr<RTCCertificate> certificate =
- RTCCertificate::Create(identity.Pass());
+ RTCCertificate::Create(std::move(identity));
uint64_t now = NowSeconds();
EXPECT_FALSE(HasExpiredSeconds(certificate, now));
diff --git a/webrtc/base/scoped_ptr.h b/webrtc/base/scoped_ptr.h
index 203a001..3b3301f 100644
--- a/webrtc/base/scoped_ptr.h
+++ b/webrtc/base/scoped_ptr.h
@@ -42,55 +42,39 @@
// }
//
// These scopers also implement part of the functionality of C++11 unique_ptr
-// in that they are "movable but not copyable." You can use the scopers in
-// the parameter and return types of functions to signify ownership transfer
-// in to and out of a function. When calling a function that has a scoper
-// as the argument type, it must be called with the result of an analogous
-// scoper's Pass() function or another function that generates a temporary;
-// passing by copy will NOT work. Here is an example using scoped_ptr:
+// in that they are "movable but not copyable." You can use the scopers in the
+// parameter and return types of functions to signify ownership transfer in to
+// and out of a function. When calling a function that has a scoper as the
+// argument type, it must be called with the result of calling std::move on an
+// analogous scoper, or another function that generates a temporary; passing by
+// copy will NOT work. Here is an example using scoped_ptr:
//
// void TakesOwnership(scoped_ptr<Foo> arg) {
// // Do something with arg
// }
// scoped_ptr<Foo> CreateFoo() {
-// // No need for calling Pass() because we are constructing a temporary
+// // No need for calling std::move because we are constructing a temporary
// // for the return value.
// return scoped_ptr<Foo>(new Foo("new"));
// }
// scoped_ptr<Foo> PassThru(scoped_ptr<Foo> arg) {
-// return arg.Pass();
+// return std::move(arg);
// }
//
// {
// scoped_ptr<Foo> ptr(new Foo("yay")); // ptr manages Foo("yay").
-// TakesOwnership(ptr.Pass()); // ptr no longer owns Foo("yay").
+// TakesOwnership(std::move(ptr)); // ptr no longer owns Foo("yay").
// scoped_ptr<Foo> ptr2 = CreateFoo(); // ptr2 owns the return Foo.
// scoped_ptr<Foo> ptr3 = // ptr3 now owns what was in ptr2.
-// PassThru(ptr2.Pass()); // ptr2 is correspondingly nullptr.
+// PassThru(std::move(ptr2)); // ptr2 is correspondingly nullptr.
// }
//
-// Notice that if you do not call Pass() when returning from PassThru(), or
+// Notice that if you do not call std::move when returning from PassThru(), or
// when invoking TakesOwnership(), the code will not compile because scopers
// are not copyable; they only implement move semantics which require calling
-// the Pass() function to signify a destructive transfer of state. CreateFoo()
-// is different though because we are constructing a temporary on the return
-// line and thus can avoid needing to call Pass().
-//
-// Pass() properly handles upcast in initialization, i.e. you can use a
-// scoped_ptr<Child> to initialize a scoped_ptr<Parent>:
-//
-// scoped_ptr<Foo> foo(new Foo());
-// scoped_ptr<FooParent> parent(foo.Pass());
-//
-// PassAs<>() should be used to upcast return value in return statement:
-//
-// scoped_ptr<Foo> CreateFoo() {
-// scoped_ptr<FooChild> result(new FooChild());
-// return result.PassAs<Foo>();
-// }
-//
-// Note that PassAs<>() is implemented only for scoped_ptr<T>, but not for
-// scoped_ptr<T[]>. This is because casting array pointers may not be safe.
+// std::move to signify a destructive transfer of state. CreateFoo() is
+// different though because we are constructing a temporary on the return line
+// and thus can avoid needing to call std::move.
#ifndef WEBRTC_BASE_SCOPED_PTR_H__
#define WEBRTC_BASE_SCOPED_PTR_H__
diff --git a/webrtc/common_audio/audio_converter.cc b/webrtc/common_audio/audio_converter.cc
index f1709ae..9bb5895 100644
--- a/webrtc/common_audio/audio_converter.cc
+++ b/webrtc/common_audio/audio_converter.cc
@@ -11,6 +11,7 @@
#include "webrtc/common_audio/audio_converter.h"
#include <cstring>
+#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/base/safe_conversions.h"
@@ -105,7 +106,7 @@
class CompositionConverter : public AudioConverter {
public:
CompositionConverter(ScopedVector<AudioConverter> converters)
- : converters_(converters.Pass()) {
+ : converters_(std::move(converters)) {
RTC_CHECK_GE(converters_.size(), 2u);
// We need an intermediate buffer after every converter.
for (auto it = converters_.begin(); it != converters_.end() - 1; ++it)
@@ -147,7 +148,7 @@
dst_channels, src_frames));
converters.push_back(new ResampleConverter(dst_channels, src_frames,
dst_channels, dst_frames));
- sp.reset(new CompositionConverter(converters.Pass()));
+ sp.reset(new CompositionConverter(std::move(converters)));
} else {
sp.reset(new DownmixConverter(src_channels, src_frames, dst_channels,
dst_frames));
@@ -159,7 +160,7 @@
src_channels, dst_frames));
converters.push_back(new UpmixConverter(src_channels, dst_frames,
dst_channels, dst_frames));
- sp.reset(new CompositionConverter(converters.Pass()));
+ sp.reset(new CompositionConverter(std::move(converters)));
} else {
sp.reset(new UpmixConverter(src_channels, src_frames, dst_channels,
dst_frames));
@@ -172,7 +173,7 @@
dst_frames));
}
- return sp.Pass();
+ return sp;
}
// For CompositionConverter.
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
index e0d1b6f..14302e4 100644
--- a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
@@ -10,6 +10,8 @@
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include <utility>
+
#include "webrtc/base/logging.h"
#include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
#include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
@@ -235,7 +237,7 @@
CreateEncoder(codec_inst, &isac_bandwidth_info_);
if (!enc)
return nullptr;
- speech_encoder_ = enc.Pass();
+ speech_encoder_ = std::move(enc);
return speech_encoder_.get();
}
diff --git a/webrtc/modules/audio_device/android/audio_manager.cc b/webrtc/modules/audio_device/android/audio_manager.cc
index 522010e..5cca52d 100644
--- a/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/webrtc/modules/audio_device/android/audio_manager.cc
@@ -10,6 +10,8 @@
#include "webrtc/modules/audio_device/android/audio_manager.h"
+#include <utility>
+
#include <android/log.h>
#include "webrtc/base/arraysize.h"
@@ -29,15 +31,16 @@
// AudioManager::JavaAudioManager implementation
AudioManager::JavaAudioManager::JavaAudioManager(
- NativeRegistration* native_reg, rtc::scoped_ptr<GlobalRef> audio_manager)
- : audio_manager_(audio_manager.Pass()),
+ NativeRegistration* native_reg,
+ rtc::scoped_ptr<GlobalRef> audio_manager)
+ : audio_manager_(std::move(audio_manager)),
init_(native_reg->GetMethodId("init", "()Z")),
dispose_(native_reg->GetMethodId("dispose", "()V")),
is_communication_mode_enabled_(
native_reg->GetMethodId("isCommunicationModeEnabled", "()Z")),
is_device_blacklisted_for_open_sles_usage_(
- native_reg->GetMethodId(
- "isDeviceBlacklistedForOpenSLESUsage", "()Z")) {
+ native_reg->GetMethodId("isDeviceBlacklistedForOpenSLESUsage",
+ "()Z")) {
ALOGD("JavaAudioManager::ctor%s", GetThreadInfo().c_str());
}
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.cc b/webrtc/modules/audio_device/android/audio_record_jni.cc
index ba3212a..4a63197 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.cc
+++ b/webrtc/modules/audio_device/android/audio_record_jni.cc
@@ -10,6 +10,8 @@
#include "webrtc/modules/audio_device/android/audio_record_jni.h"
+#include <utility>
+
#include <android/log.h>
#include "webrtc/base/arraysize.h"
@@ -28,18 +30,15 @@
// AudioRecordJni::JavaAudioRecord implementation.
AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
- NativeRegistration* native_reg, rtc::scoped_ptr<GlobalRef> audio_record)
- : audio_record_(audio_record.Pass()),
+ NativeRegistration* native_reg,
+ rtc::scoped_ptr<GlobalRef> audio_record)
+ : audio_record_(std::move(audio_record)),
init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
- enable_built_in_aec_(native_reg->GetMethodId(
- "enableBuiltInAEC", "(Z)Z")),
- enable_built_in_agc_(native_reg->GetMethodId(
- "enableBuiltInAGC", "(Z)Z")),
- enable_built_in_ns_(native_reg->GetMethodId(
- "enableBuiltInNS", "(Z)Z")) {
-}
+ enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
+ enable_built_in_agc_(native_reg->GetMethodId("enableBuiltInAGC", "(Z)Z")),
+ enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
diff --git a/webrtc/modules/audio_device/android/audio_track_jni.cc b/webrtc/modules/audio_device/android/audio_track_jni.cc
index 29b21ae..c660868 100644
--- a/webrtc/modules/audio_device/android/audio_track_jni.cc
+++ b/webrtc/modules/audio_device/android/audio_track_jni.cc
@@ -11,6 +11,8 @@
#include "webrtc/modules/audio_device/android/audio_manager.h"
#include "webrtc/modules/audio_device/android/audio_track_jni.h"
+#include <utility>
+
#include <android/log.h>
#include "webrtc/base/arraysize.h"
@@ -28,16 +30,16 @@
// AudioTrackJni::JavaAudioTrack implementation.
AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
- NativeRegistration* native_reg, rtc::scoped_ptr<GlobalRef> audio_track)
- : audio_track_(audio_track.Pass()),
+ NativeRegistration* native_reg,
+ rtc::scoped_ptr<GlobalRef> audio_track)
+ : audio_track_(std::move(audio_track)),
init_playout_(native_reg->GetMethodId("initPlayout", "(II)V")),
start_playout_(native_reg->GetMethodId("startPlayout", "()Z")),
stop_playout_(native_reg->GetMethodId("stopPlayout", "()Z")),
set_stream_volume_(native_reg->GetMethodId("setStreamVolume", "(I)Z")),
- get_stream_max_volume_(native_reg->GetMethodId(
- "getStreamMaxVolume", "()I")),
- get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")) {
-}
+ get_stream_max_volume_(
+ native_reg->GetMethodId("getStreamMaxVolume", "()I")),
+ get_stream_volume_(native_reg->GetMethodId("getStreamVolume", "()I")) {}
AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
diff --git a/webrtc/modules/audio_processing/test/audio_file_processor.cc b/webrtc/modules/audio_processing/test/audio_file_processor.cc
index ca244d5..4c77356 100644
--- a/webrtc/modules/audio_processing/test/audio_file_processor.cc
+++ b/webrtc/modules/audio_processing/test/audio_file_processor.cc
@@ -11,6 +11,7 @@
#include "webrtc/modules/audio_processing/test/audio_file_processor.h"
#include <algorithm>
+#include <utility>
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_processing/test/protobuf_utils.h"
@@ -43,13 +44,13 @@
WavFileProcessor::WavFileProcessor(scoped_ptr<AudioProcessing> ap,
scoped_ptr<WavReader> in_file,
scoped_ptr<WavWriter> out_file)
- : ap_(ap.Pass()),
+ : ap_(std::move(ap)),
in_buf_(GetChannelBuffer(*in_file)),
out_buf_(GetChannelBuffer(*out_file)),
input_config_(GetStreamConfig(*in_file)),
output_config_(GetStreamConfig(*out_file)),
- buffer_reader_(in_file.Pass()),
- buffer_writer_(out_file.Pass()) {}
+ buffer_reader_(std::move(in_file)),
+ buffer_writer_(std::move(out_file)) {}
bool WavFileProcessor::ProcessChunk() {
if (!buffer_reader_.Read(&in_buf_)) {
@@ -68,11 +69,11 @@
AecDumpFileProcessor::AecDumpFileProcessor(scoped_ptr<AudioProcessing> ap,
FILE* dump_file,
scoped_ptr<WavWriter> out_file)
- : ap_(ap.Pass()),
+ : ap_(std::move(ap)),
dump_file_(dump_file),
out_buf_(GetChannelBuffer(*out_file)),
output_config_(GetStreamConfig(*out_file)),
- buffer_writer_(out_file.Pass()) {
+ buffer_writer_(std::move(out_file)) {
RTC_CHECK(dump_file_) << "Could not open dump file for reading.";
}
diff --git a/webrtc/modules/audio_processing/test/audioproc_float.cc b/webrtc/modules/audio_processing/test/audioproc_float.cc
index 3f1dc37..d64b006 100644
--- a/webrtc/modules/audio_processing/test/audioproc_float.cc
+++ b/webrtc/modules/audio_processing/test/audioproc_float.cc
@@ -12,6 +12,7 @@
#include <iostream>
#include <sstream>
#include <string>
+#include <utility>
#include "gflags/gflags.h"
#include "webrtc/base/checks.h"
@@ -122,12 +123,12 @@
if (FLAGS_dump.empty()) {
auto in_file = rtc_make_scoped_ptr(new WavReader(FLAGS_i));
std::cout << FLAGS_i << ": " << in_file->FormatAsString() << std::endl;
- processor.reset(
- new WavFileProcessor(ap.Pass(), in_file.Pass(), out_file.Pass()));
+ processor.reset(new WavFileProcessor(std::move(ap), std::move(in_file),
+ std::move(out_file)));
} else {
processor.reset(new AecDumpFileProcessor(
- ap.Pass(), fopen(FLAGS_dump.c_str(), "rb"), out_file.Pass()));
+ std::move(ap), fopen(FLAGS_dump.c_str(), "rb"), std::move(out_file)));
}
int num_chunks = 0;
diff --git a/webrtc/modules/audio_processing/test/test_utils.cc b/webrtc/modules/audio_processing/test/test_utils.cc
index 47bd314..25181a1 100644
--- a/webrtc/modules/audio_processing/test/test_utils.cc
+++ b/webrtc/modules/audio_processing/test/test_utils.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <utility>
+
#include "webrtc/base/checks.h"
#include "webrtc/modules/audio_processing/test/test_utils.h"
@@ -32,7 +34,7 @@
}
ChannelBufferWavReader::ChannelBufferWavReader(rtc::scoped_ptr<WavReader> file)
- : file_(file.Pass()) {}
+ : file_(std::move(file)) {}
bool ChannelBufferWavReader::Read(ChannelBuffer<float>* buffer) {
RTC_CHECK_EQ(file_->num_channels(), buffer->num_channels());
@@ -49,7 +51,7 @@
}
ChannelBufferWavWriter::ChannelBufferWavWriter(rtc::scoped_ptr<WavWriter> file)
- : file_(file.Pass()) {}
+ : file_(std::move(file)) {}
void ChannelBufferWavWriter::Write(const ChannelBuffer<float>& buffer) {
RTC_CHECK_EQ(file_->num_channels(), buffer.num_channels());
diff --git a/webrtc/modules/desktop_capture/screen_capturer_win.cc b/webrtc/modules/desktop_capture/screen_capturer_win.cc
index 1f33155..18be4eb 100644
--- a/webrtc/modules/desktop_capture/screen_capturer_win.cc
+++ b/webrtc/modules/desktop_capture/screen_capturer_win.cc
@@ -10,6 +10,8 @@
#include "webrtc/modules/desktop_capture/screen_capturer.h"
+#include <utility>
+
#include "webrtc/modules/desktop_capture/desktop_capture_options.h"
#include "webrtc/modules/desktop_capture/win/screen_capturer_win_gdi.h"
#include "webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.h"
@@ -22,7 +24,7 @@
new ScreenCapturerWinGdi(options));
if (options.allow_use_magnification_api())
- return new ScreenCapturerWinMagnifier(gdi_capturer.Pass());
+ return new ScreenCapturerWinMagnifier(std::move(gdi_capturer));
return gdi_capturer.release();
}
diff --git a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
index b5eb1c0..066943d 100644
--- a/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
+++ b/webrtc/modules/desktop_capture/win/screen_capturer_win_magnifier.cc
@@ -12,6 +12,8 @@
#include <assert.h>
+#include <utility>
+
#include "webrtc/modules/desktop_capture/desktop_capture_options.h"
#include "webrtc/modules/desktop_capture/desktop_frame.h"
#include "webrtc/modules/desktop_capture/desktop_frame_win.h"
@@ -37,7 +39,7 @@
ScreenCapturerWinMagnifier::ScreenCapturerWinMagnifier(
rtc::scoped_ptr<ScreenCapturer> fallback_capturer)
- : fallback_capturer_(fallback_capturer.Pass()),
+ : fallback_capturer_(std::move(fallback_capturer)),
fallback_capturer_started_(false),
callback_(NULL),
current_screen_id_(kFullDesktopScreenId),
@@ -53,8 +55,7 @@
host_window_(NULL),
magnifier_window_(NULL),
magnifier_initialized_(false),
- magnifier_capture_succeeded_(true) {
-}
+ magnifier_capture_succeeded_(true) {}
ScreenCapturerWinMagnifier::~ScreenCapturerWinMagnifier() {
// DestroyWindow must be called before MagUninitialize. magnifier_window_ is
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
index 01ded72..4451fe8 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/app_unittest.cc
@@ -29,7 +29,7 @@
class RtcpPacketAppTest : public ::testing::Test {
protected:
- void BuildPacket() { packet = app.Build().Pass(); }
+ void BuildPacket() { packet = app.Build(); }
void ParsePacket() {
RtcpCommonHeader header;
EXPECT_TRUE(
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
index 0728ed9..d2ae8ed 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/bye_unittest.cc
@@ -30,7 +30,7 @@
class RtcpPacketByeTest : public ::testing::Test {
protected:
- void BuildPacket() { packet = bye.Build().Pass(); }
+ void BuildPacket() { packet = bye.Build(); }
void ParsePacket() {
RtcpCommonHeader header;
EXPECT_TRUE(
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
index 673e1d8..09d7b63 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/extended_jitter_report_unittest.cc
@@ -25,7 +25,7 @@
class RtcpPacketExtendedJitterReportTest : public ::testing::Test {
protected:
- void BuildPacket() { packet = ij.Build().Pass(); }
+ void BuildPacket() { packet = ij.Build(); }
void ParsePacket() {
RtcpCommonHeader header;
EXPECT_TRUE(
diff --git a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
index 4fd329f..ff3da60 100644
--- a/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtcp_packet/receiver_report_unittest.cc
@@ -39,7 +39,7 @@
class RtcpPacketReceiverReportTest : public ::testing::Test {
protected:
- void BuildPacket() { packet = rr.Build().Pass(); }
+ void BuildPacket() { packet = rr.Build(); }
void ParsePacket() {
RtcpCommonHeader header;
EXPECT_TRUE(
diff --git a/webrtc/modules/utility/source/process_thread_impl.cc b/webrtc/modules/utility/source/process_thread_impl.cc
index 1da4225..8cdf016 100644
--- a/webrtc/modules/utility/source/process_thread_impl.cc
+++ b/webrtc/modules/utility/source/process_thread_impl.cc
@@ -38,8 +38,7 @@
// static
rtc::scoped_ptr<ProcessThread> ProcessThread::Create(
const char* thread_name) {
- return rtc::scoped_ptr<ProcessThread>(new ProcessThreadImpl(thread_name))
- .Pass();
+ return rtc::scoped_ptr<ProcessThread>(new ProcessThreadImpl(thread_name));
}
ProcessThreadImpl::ProcessThreadImpl(const char* thread_name)
diff --git a/webrtc/modules/utility/source/process_thread_impl_unittest.cc b/webrtc/modules/utility/source/process_thread_impl_unittest.cc
index 92cd561..0b35fad 100644
--- a/webrtc/modules/utility/source/process_thread_impl_unittest.cc
+++ b/webrtc/modules/utility/source/process_thread_impl_unittest.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <utility>
+
#include "testing/gmock/include/gmock/gmock.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/include/module.h"
@@ -295,7 +297,7 @@
rtc::scoped_ptr<EventWrapper> task_ran(EventWrapper::Create());
rtc::scoped_ptr<RaiseEventTask> task(new RaiseEventTask(task_ran.get()));
thread.Start();
- thread.PostTask(task.Pass());
+ thread.PostTask(std::move(task));
EXPECT_EQ(kEventSignaled, task_ran->Wait(100));
thread.Stop();
}
diff --git a/webrtc/modules/video_coding/jitter_buffer.cc b/webrtc/modules/video_coding/jitter_buffer.cc
index d44d2b6..a1142bb 100644
--- a/webrtc/modules/video_coding/jitter_buffer.cc
+++ b/webrtc/modules/video_coding/jitter_buffer.cc
@@ -219,7 +219,7 @@
: clock_(clock),
running_(false),
crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
- frame_event_(event.Pass()),
+ frame_event_(std::move(event)),
max_number_of_frames_(kStartNumberOfFrames),
free_frames_(),
decodable_frames_(),
diff --git a/webrtc/modules/video_coding/receiver.cc b/webrtc/modules/video_coding/receiver.cc
index fe05b86..91cdd5e 100644
--- a/webrtc/modules/video_coding/receiver.cc
+++ b/webrtc/modules/video_coding/receiver.cc
@@ -13,6 +13,7 @@
#include <assert.h>
#include <cstdlib>
+#include <utility>
#include "webrtc/base/logging.h"
#include "webrtc/base/trace_event.h"
@@ -40,9 +41,9 @@
rtc::scoped_ptr<EventWrapper> jitter_buffer_event)
: crit_sect_(CriticalSectionWrapper::CreateCriticalSection()),
clock_(clock),
- jitter_buffer_(clock_, jitter_buffer_event.Pass()),
+ jitter_buffer_(clock_, std::move(jitter_buffer_event)),
timing_(timing),
- render_wait_event_(receiver_event.Pass()),
+ render_wait_event_(std::move(receiver_event)),
max_video_delay_ms_(kMaxVideoDelayMs) {
Reset();
}
diff --git a/webrtc/p2p/base/dtlstransportchannel.cc b/webrtc/p2p/base/dtlstransportchannel.cc
index 95014f9..e1de451 100644
--- a/webrtc/p2p/base/dtlstransportchannel.cc
+++ b/webrtc/p2p/base/dtlstransportchannel.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <utility>
+
#include "webrtc/p2p/base/dtlstransportchannel.h"
#include "webrtc/p2p/base/common.h"
@@ -224,7 +226,7 @@
}
// At this point we know we are doing DTLS
- remote_fingerprint_value_ = remote_fingerprint_value.Pass();
+ remote_fingerprint_value_ = std::move(remote_fingerprint_value);
remote_fingerprint_algorithm_ = digest_alg;
bool reconnect = dtls_;
diff --git a/webrtc/p2p/base/dtlstransportchannel_unittest.cc b/webrtc/p2p/base/dtlstransportchannel_unittest.cc
index a967ae3..c2cae73 100644
--- a/webrtc/p2p/base/dtlstransportchannel_unittest.cc
+++ b/webrtc/p2p/base/dtlstransportchannel_unittest.cc
@@ -53,9 +53,9 @@
received_dtls_client_hello_(false),
received_dtls_server_hello_(false) {}
void CreateCertificate(rtc::KeyType key_type) {
- certificate_ = rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- rtc::SSLIdentity::Generate(name_, key_type)).Pass());
+ certificate_ =
+ rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+ rtc::SSLIdentity::Generate(name_, key_type)));
}
const rtc::scoped_refptr<rtc::RTCCertificate>& certificate() {
return certificate_;
diff --git a/webrtc/p2p/base/transportcontroller_unittest.cc b/webrtc/p2p/base/transportcontroller_unittest.cc
index 23e4dc8..1408529 100644
--- a/webrtc/p2p/base/transportcontroller_unittest.cc
+++ b/webrtc/p2p/base/transportcontroller_unittest.cc
@@ -268,15 +268,11 @@
TEST_F(TransportControllerTest, TestSetAndGetLocalCertificate) {
rtc::scoped_refptr<rtc::RTCCertificate> certificate1 =
- rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT))
- .Pass());
+ rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+ rtc::SSLIdentity::Generate("session1", rtc::KT_DEFAULT)));
rtc::scoped_refptr<rtc::RTCCertificate> certificate2 =
- rtc::RTCCertificate::Create(
- rtc::scoped_ptr<rtc::SSLIdentity>(
- rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT))
- .Pass());
+ rtc::RTCCertificate::Create(rtc::scoped_ptr<rtc::SSLIdentity>(
+ rtc::SSLIdentity::Generate("session2", rtc::KT_DEFAULT)));
rtc::scoped_refptr<rtc::RTCCertificate> returned_certificate;
FakeTransportChannel* channel1 = CreateChannel("audio", 1);
diff --git a/webrtc/p2p/base/transportdescriptionfactory_unittest.cc b/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
index e3992df..a52d9ed 100644
--- a/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
+++ b/webrtc/p2p/base/transportdescriptionfactory_unittest.cc
@@ -26,11 +26,10 @@
class TransportDescriptionFactoryTest : public testing::Test {
public:
TransportDescriptionFactoryTest()
- : cert1_(rtc::RTCCertificate::Create(scoped_ptr<rtc::SSLIdentity>(
- new rtc::FakeSSLIdentity("User1")).Pass())),
- cert2_(rtc::RTCCertificate::Create(scoped_ptr<rtc::SSLIdentity>(
- new rtc::FakeSSLIdentity("User2")).Pass())) {
- }
+ : cert1_(rtc::RTCCertificate::Create(
+ scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("User1")))),
+ cert2_(rtc::RTCCertificate::Create(
+ scoped_ptr<rtc::SSLIdentity>(new rtc::FakeSSLIdentity("User2")))) {}
void CheckDesc(const TransportDescription* desc,
const std::string& opt, const std::string& ice_ufrag,
