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Unified Diff: webrtc/api/webrtcsession.h

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.h
diff --git a/webrtc/api/webrtcsession.h b/webrtc/api/webrtcsession.h
index 08938fc9f61635a6c23382d1271ef2f41146aa65..89b77bbc884b34950b83fc78ea45b050690fadb4 100644
--- a/webrtc/api/webrtcsession.h
+++ b/webrtc/api/webrtcsession.h
@@ -153,7 +153,7 @@ class WebRtcSession : public AudioProviderInterface,
bool Initialize(
const PeerConnectionFactoryInterface::Options& options,
- rtc::scoped_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
+ std::unique_ptr<DtlsIdentityStoreInterface> dtls_identity_store,
const PeerConnectionInterface::RTCConfiguration& rtc_configuration);
// Deletes the voice, video and data channel and changes the session state
// to STATE_CLOSED.
@@ -244,7 +244,7 @@ class WebRtcSession : public AudioProviderInterface,
cricket::AudioSource* source) override;
void SetAudioPlayoutVolume(uint32_t ssrc, double volume) override;
void SetRawAudioSink(uint32_t ssrc,
- rtc::scoped_ptr<AudioSinkInterface> sink) override;
+ std::unique_ptr<AudioSinkInterface> sink) override;
RtpParameters GetAudioRtpParameters(uint32_t ssrc) const override;
bool SetAudioRtpParameters(uint32_t ssrc,
@@ -479,17 +479,17 @@ class WebRtcSession : public AudioProviderInterface,
const std::string sid_;
bool initial_offerer_ = false;
- rtc::scoped_ptr<cricket::TransportController> transport_controller_;
+ std::unique_ptr<cricket::TransportController> transport_controller_;
MediaControllerInterface* media_controller_;
- rtc::scoped_ptr<cricket::VoiceChannel> voice_channel_;
- rtc::scoped_ptr<cricket::VideoChannel> video_channel_;
- rtc::scoped_ptr<cricket::DataChannel> data_channel_;
+ std::unique_ptr<cricket::VoiceChannel> voice_channel_;
+ std::unique_ptr<cricket::VideoChannel> video_channel_;
+ std::unique_ptr<cricket::DataChannel> data_channel_;
cricket::ChannelManager* channel_manager_;
IceObserver* ice_observer_;
PeerConnectionInterface::IceConnectionState ice_connection_state_;
bool ice_connection_receiving_;
- rtc::scoped_ptr<SessionDescriptionInterface> local_desc_;
- rtc::scoped_ptr<SessionDescriptionInterface> remote_desc_;
+ std::unique_ptr<SessionDescriptionInterface> local_desc_;
+ std::unique_ptr<SessionDescriptionInterface> remote_desc_;
// If the remote peer is using a older version of implementation.
bool older_version_remote_peer_;
bool dtls_enabled_;
@@ -504,8 +504,7 @@ class WebRtcSession : public AudioProviderInterface,
// List of content names for which the remote side triggered an ICE restart.
std::set<std::string> pending_ice_restarts_;
- rtc::scoped_ptr<WebRtcSessionDescriptionFactory>
- webrtc_session_desc_factory_;
+ std::unique_ptr<WebRtcSessionDescriptionFactory> webrtc_session_desc_factory_;
// Member variables for caching global options.
cricket::AudioOptions audio_options_;
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