Replace scoped_ptr with unique_ptr in webrtc/modules/audio_device/

webrtc/modules/audio_device/android/audio_record_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 {
 
 // AudioRecordJni::JavaAudioRecord implementation.
 AudioRecordJni::JavaAudioRecord::JavaAudioRecord(
     NativeRegistration* native_reg,
-    rtc::scoped_ptr<GlobalRef> audio_record)
+    std::unique_ptr<GlobalRef> audio_record)
     : audio_record_(std::move(audio_record)),
       init_recording_(native_reg->GetMethodId("initRecording", "(II)I")),
       start_recording_(native_reg->GetMethodId("startRecording", "()Z")),
       stop_recording_(native_reg->GetMethodId("stopRecording", "()Z")),
       enable_built_in_aec_(native_reg->GetMethodId("enableBuiltInAEC", "(Z)Z")),
       enable_built_in_agc_(native_reg->GetMethodId("enableBuiltInAGC", "(Z)Z")),
       enable_built_in_ns_(native_reg->GetMethodId("enableBuiltInNS", "(Z)Z")) {}
 
 AudioRecordJni::JavaAudioRecord::~JavaAudioRecord() {}
 
@@ skipping 22 matching lines @@
                                           static_cast<jboolean>(enable));
 }
 
 bool AudioRecordJni::JavaAudioRecord::EnableBuiltInNS(bool enable) {
   return audio_record_->CallBooleanMethod(enable_built_in_ns_,
                                           static_cast<jboolean>(enable));
 }
 
 // AudioRecordJni implementation.
 AudioRecordJni::AudioRecordJni(AudioManager* audio_manager)
-    : j_environment_(JVM::GetInstance()->environment()),
+    : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())),
       audio_manager_(audio_manager),
       audio_parameters_(audio_manager->GetRecordAudioParameters()),
       total_delay_in_milliseconds_(0),
       direct_buffer_address_(nullptr),
       direct_buffer_capacity_in_bytes_(0),
       frames_per_buffer_(0),
       initialized_(false),
       recording_(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::AudioRecordJni::CacheDirectBufferAddress)},
       {"nativeDataIsRecorded", "(IJ)V",
       reinterpret_cast<void*>(&webrtc::AudioRecordJni::DataIsRecorded)}};
-  j_native_registration_ = j_environment_->RegisterNatives(
+  j_native_registration_ = rtc::ScopedToUnique(j_environment_->RegisterNatives(
       "org/webrtc/voiceengine/WebRtcAudioRecord",
-      native_methods, arraysize(native_methods));
+      native_methods, arraysize(native_methods)));
   j_audio_record_.reset(new JavaAudioRecord(
       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();
 }
 
 AudioRecordJni::~AudioRecordJni() {
   ALOGD("~dtor%s", GetThreadInfo().c_str());
   RTC_DCHECK(thread_checker_.CalledOnValidThread());
   Terminate();
 }
@@ skipping 139 matching lines @@
   // of |playDelayMs| and |recDelayMs|, hence the distributions does not matter.
   audio_device_buffer_->SetVQEData(total_delay_in_milliseconds_,
                                    0,   // recDelayMs
                                    0);  // clockDrift
   if (audio_device_buffer_->DeliverRecordedData() == -1) {
     ALOGE("AudioDeviceBuffer::DeliverRecordedData failed!");
   }
 }
 
 }  // namespace webrtc