Replace scoped_ptr with unique_ptr in webrtc/audio/

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1706183002

Cr-Commit-Position: refs/heads/master@{#11723}

webrtc/audio/audio_receive_stream.cc

@@ -93,8 +93,7 @@ AudioReceiveStream::AudioReceiveStream(
   RTC_DCHECK(rtp_header_parser_);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
-  channel_proxy_ =
-      rtc::UniqueToScoped(voe_impl->GetChannelProxy(config_.voe_channel_id));
+  channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->SetLocalSSRC(config.rtp.local_ssrc);
   for (const auto& extension : config.rtp.extensions) {
     if (extension.name == RtpExtension::kAudioLevel) {
@@ -229,9 +228,9 @@ webrtc::AudioReceiveStream::Stats AudioReceiveStream::GetStats() const {
   return stats;
 }
 
-void AudioReceiveStream::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void AudioReceiveStream::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
-  channel_proxy_->SetSink(rtc::ScopedToUnique(std::move(sink)));
+  channel_proxy_->SetSink(std::move(sink));
 }
 
 const webrtc::AudioReceiveStream::Config& AudioReceiveStream::config() const {

webrtc/audio/audio_receive_stream.h

@@ -11,6 +11,8 @@
 #ifndef WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_RECEIVE_STREAM_H_
 
+#include <memory>
+
 #include "webrtc/audio_receive_stream.h"
 #include "webrtc/audio_state.h"
 #include "webrtc/base/thread_checker.h"
@@ -45,7 +47,7 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
   // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
 
-  void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) override;
+  void SetSink(std::unique_ptr<AudioSinkInterface> sink) override;
 
   const webrtc::AudioReceiveStream::Config& config() const;
 
@@ -56,8 +58,8 @@ class AudioReceiveStream final : public webrtc::AudioReceiveStream {
   RemoteBitrateEstimator* remote_bitrate_estimator_ = nullptr;
   const webrtc::AudioReceiveStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
-  rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
-  rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
+  std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
+  std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioReceiveStream);
 };

webrtc/audio/audio_send_stream.cc

@@ -67,8 +67,7 @@ AudioSendStream::AudioSendStream(
   RTC_DCHECK(congestion_controller);
 
   VoiceEngineImpl* voe_impl = static_cast<VoiceEngineImpl*>(voice_engine());
-  channel_proxy_ =
-      rtc::UniqueToScoped(voe_impl->GetChannelProxy(config_.voe_channel_id));
+  channel_proxy_ = voe_impl->GetChannelProxy(config_.voe_channel_id);
   channel_proxy_->RegisterSenderCongestionControlObjects(
       congestion_controller->pacer(),
       congestion_controller->GetTransportFeedbackObserver(),

webrtc/audio/audio_send_stream.h

@@ -11,10 +11,11 @@
 #ifndef WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 #define WEBRTC_AUDIO_AUDIO_SEND_STREAM_H_
 
+#include <memory>
+
 #include "webrtc/audio_send_stream.h"
 #include "webrtc/audio_state.h"
 #include "webrtc/base/thread_checker.h"
-#include "webrtc/base/scoped_ptr.h"
 
 namespace webrtc {
 class CongestionController;
@@ -51,7 +52,7 @@ class AudioSendStream final : public webrtc::AudioSendStream {
   rtc::ThreadChecker thread_checker_;
   const webrtc::AudioSendStream::Config config_;
   rtc::scoped_refptr<webrtc::AudioState> audio_state_;
-  rtc::scoped_ptr<voe::ChannelProxy> channel_proxy_;
+  std::unique_ptr<voe::ChannelProxy> channel_proxy_;
 
   RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioSendStream);
 };

webrtc/audio/audio_state_unittest.cc

@@ -8,10 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/audio/audio_state.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/test/mock_voice_engine.h"
 
 namespace webrtc {
@@ -44,20 +45,20 @@ TEST(AudioStateTest, Create) {
 
 TEST(AudioStateTest, ConstructDestruct) {
   ConfigHelper helper;
-  rtc::scoped_ptr<internal::AudioState> audio_state(
+  std::unique_ptr<internal::AudioState> audio_state(
       new internal::AudioState(helper.config()));
 }
 
 TEST(AudioStateTest, GetVoiceEngine) {
   ConfigHelper helper;
-  rtc::scoped_ptr<internal::AudioState> audio_state(
+  std::unique_ptr<internal::AudioState> audio_state(
       new internal::AudioState(helper.config()));
   EXPECT_EQ(audio_state->voice_engine(), &helper.voice_engine());
 }
 
 TEST(AudioStateTest, TypingNoiseDetected) {
   ConfigHelper helper;
-  rtc::scoped_ptr<internal::AudioState> audio_state(
+  std::unique_ptr<internal::AudioState> audio_state(
       new internal::AudioState(helper.config()));
   VoiceEngineObserver* voe_observer =
       static_cast<VoiceEngineObserver*>(audio_state.get());

webrtc/audio_receive_stream.h

@@ -12,10 +12,10 @@
 #define WEBRTC_AUDIO_RECEIVE_STREAM_H_
 
 #include <map>
+#include <memory>
 #include <string>
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/config.h"
 #include "webrtc/stream.h"
 #include "webrtc/transport.h"
@@ -114,7 +114,7 @@ class AudioReceiveStream : public ReceiveStream {
   // to stream through this sink. In practice, this happens if mixed audio
   // is being pulled+rendered and/or if audio is being pulled for the purposes
   // of feeding to the AEC.
-  virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) = 0;
+  virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink) = 0;
 };
 }  // namespace webrtc
 

webrtc/media/engine/fakewebrtccall.cc

@@ -76,8 +76,8 @@ webrtc::AudioReceiveStream::Stats FakeAudioReceiveStream::GetStats() const {
 }
 
 void FakeAudioReceiveStream::SetSink(
-    rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
-  sink_ = std::move(sink);
+    std::unique_ptr<webrtc::AudioSinkInterface> sink) {
+  sink_ = rtc::UniqueToScoped(std::move(sink));
 }
 
 FakeVideoSendStream::FakeVideoSendStream(

webrtc/media/engine/fakewebrtccall.h

@@ -20,6 +20,7 @@
 #ifndef WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_
 #define WEBRTC_MEDIA_ENGINE_FAKEWEBRTCCALL_H_
 
+#include <memory>
 #include <vector>
 
 #include "webrtc/audio_receive_stream.h"
@@ -90,7 +91,7 @@ class FakeAudioReceiveStream final : public webrtc::AudioReceiveStream {
 
   // webrtc::AudioReceiveStream implementation.
   webrtc::AudioReceiveStream::Stats GetStats() const override;
-  void SetSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) override;
+  void SetSink(std::unique_ptr<webrtc::AudioSinkInterface> sink) override;
 
   webrtc::AudioReceiveStream::Config config_;
   webrtc::AudioReceiveStream::Stats stats_;

webrtc/media/engine/webrtcvoiceengine.cc

@@ -1338,7 +1338,7 @@ class WebRtcVoiceMediaChannel::WebRtcAudioReceiveStream {
 
   void SetRawAudioSink(rtc::scoped_ptr<webrtc::AudioSinkInterface> sink) {
     RTC_DCHECK(worker_thread_checker_.CalledOnValidThread());
-    stream_->SetSink(std::move(sink));
+    stream_->SetSink(rtc::ScopedToUnique(std::move(sink)));
   }
 
  private: