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Unified Diff: webrtc/api/webrtcsession.cc

Issue 1930463002: Replace scoped_ptr with unique_ptr in webrtc/api/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 8 months ago
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Index: webrtc/api/webrtcsession.cc
diff --git a/webrtc/api/webrtcsession.cc b/webrtc/api/webrtcsession.cc
index 1cf7924fb77f3e062a9fc907df4783cfdd10589c..9f84840822fce12c4c9012b568e10c0b3b4e651b 100644
--- a/webrtc/api/webrtcsession.cc
+++ b/webrtc/api/webrtcsession.cc
@@ -528,7 +528,7 @@ WebRtcSession::~WebRtcSession() {
bool WebRtcSession::Initialize(
const PeerConnectionFactoryInterface::Options& options,
- rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+ std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
const PeerConnectionInterface::RTCConfiguration& rtc_configuration) {
bundle_policy_ = rtc_configuration.bundle_policy;
rtcp_mux_policy_ = rtc_configuration.rtcp_mux_policy;
@@ -675,7 +675,7 @@ bool WebRtcSession::SetLocalDescription(SessionDescriptionInterface* desc,
ASSERT(signaling_thread()->IsCurrent());
// Takes the ownership of |desc| regardless of the result.
- rtc::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);
+ std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
// Validate SDP.
if (!ValidateSessionDescription(desc, cricket::CS_LOCAL, err_desc)) {
@@ -731,14 +731,14 @@ bool WebRtcSession::SetRemoteDescription(SessionDescriptionInterface* desc,
ASSERT(signaling_thread()->IsCurrent());
// Takes the ownership of |desc| regardless of the result.
- rtc::scoped_ptr<SessionDescriptionInterface> desc_temp(desc);
+ std::unique_ptr<SessionDescriptionInterface> desc_temp(desc);
// Validate SDP.
if (!ValidateSessionDescription(desc, cricket::CS_REMOTE, err_desc)) {
return false;
}
- rtc::scoped_ptr<SessionDescriptionInterface> old_remote_desc(
+ std::unique_ptr<SessionDescriptionInterface> old_remote_desc(
remote_desc_.release());
remote_desc_.reset(desc_temp.release());
@@ -1236,7 +1236,7 @@ void WebRtcSession::SetAudioPlayoutVolume(uint32_t ssrc, double volume) {
}
void WebRtcSession::SetRawAudioSink(uint32_t ssrc,
- rtc::scoped_ptr<AudioSinkInterface> sink) {
+ std::unique_ptr<AudioSinkInterface> sink) {
ASSERT(signaling_thread()->IsCurrent());
if (!voice_channel_)
return;
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