Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/

webrtc/modules/audio_device/android/audio_track_jni.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 13 matching lines @@
 #define ALOGD(...) __android_log_print(ANDROID_LOG_DEBUG, TAG, __VA_ARGS__)
 #define ALOGE(...) __android_log_print(ANDROID_LOG_ERROR, TAG, __VA_ARGS__)
 #define ALOGW(...) __android_log_print(ANDROID_LOG_WARN, TAG, __VA_ARGS__)
 #define ALOGI(...) __android_log_print(ANDROID_LOG_INFO, TAG, __VA_ARGS__)
 
 namespace webrtc {
 
 // AudioTrackJni::JavaAudioTrack implementation.
 AudioTrackJni::JavaAudioTrack::JavaAudioTrack(
     NativeRegistration* native_reg,
-    rtc::scoped_ptr<GlobalRef> audio_track)
+    std::unique_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")) {}
 
 AudioTrackJni::JavaAudioTrack::~JavaAudioTrack() {}
@@ skipping 17 matching lines @@
 int AudioTrackJni::JavaAudioTrack::GetStreamMaxVolume() {
   return audio_track_->CallIntMethod(get_stream_max_volume_);
 }
 
 int AudioTrackJni::JavaAudioTrack::GetStreamVolume() {
   return audio_track_->CallIntMethod(get_stream_volume_);
 }
 
 // TODO(henrika): possible extend usage of AudioManager and add it as member.
 AudioTrackJni::AudioTrackJni(AudioManager* audio_manager)
-    : j_environment_(JVM::GetInstance()->environment()),
+    : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())),
       audio_parameters_(audio_manager->GetPlayoutAudioParameters()),
       direct_buffer_address_(nullptr),
       direct_buffer_capacity_in_bytes_(0),
       frames_per_buffer_(0),
       initialized_(false),
       playing_(false),
       audio_device_buffer_(nullptr) {
   ALOGD("ctor%s", GetThreadInfo().c_str());
   RTC_DCHECK(audio_parameters_.is_valid());
   RTC_CHECK(j_environment_);
   JNINativeMethod native_methods[] = {
       {"nativeCacheDirectBufferAddress", "(Ljava/nio/ByteBuffer;J)V",
       reinterpret_cast<void*>(
           &webrtc::AudioTrackJni::CacheDirectBufferAddress)},
       {"nativeGetPlayoutData", "(IJ)V",
       reinterpret_cast<void*>(&webrtc::AudioTrackJni::GetPlayoutData)}};
-  j_native_registration_ = j_environment_->RegisterNatives(
+  j_native_registration_ = rtc::ScopedToUnique(j_environment_->RegisterNatives(
       "org/webrtc/voiceengine/WebRtcAudioTrack",
-      native_methods, arraysize(native_methods));
+      native_methods, arraysize(native_methods)));
   j_audio_track_.reset(new JavaAudioTrack(
       j_native_registration_.get(),
-      j_native_registration_->NewObject(
+      rtc::ScopedToUnique(j_native_registration_->NewObject(
           "<init>", "(Landroid/content/Context;J)V",
-          JVM::GetInstance()->context(), PointerTojlong(this))));
+          JVM::GetInstance()->context(), PointerTojlong(this)))));
   // Detach from this thread since we want to use the checker to verify calls
   // from the Java based audio thread.
   thread_checker_java_.DetachFromThread();
 }
 
 AudioTrackJni::~AudioTrackJni() {
   ALOGD("~dtor%s", GetThreadInfo().c_str());
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   Terminate();
 }
@@ skipping 144 matching lines @@
     return;
   }
   RTC_DCHECK_EQ(static_cast<size_t>(samples), frames_per_buffer_);
   // Copy decoded data into common byte buffer to ensure that it can be
   // written to the Java based audio track.
   samples = audio_device_buffer_->GetPlayoutData(direct_buffer_address_);
   RTC_DCHECK_EQ(length, kBytesPerFrame * samples);
 }
 
 }  // namespace webrtc