| 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;
|
|
|