diff --git a/webrtc/modules/audio_device/android/audio_device_unittest.cc b/webrtc/modules/audio_device/android/audio_device_unittest.cc
index 768047df51..7655c820ad 100644
--- a/webrtc/modules/audio_device/android/audio_device_unittest.cc
+++ b/webrtc/modules/audio_device/android/audio_device_unittest.cc
@@ -11,6 +11,7 @@
 #include <algorithm>
 #include <limits>
 #include <list>
+#include <memory>
 #include <numeric>
 #include <string>
 #include <vector>
@@ -20,7 +21,6 @@
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
@@ -144,7 +144,7 @@ class FileAudioStream : public AudioStreamInterface {
  private:
   size_t file_size_in_bytes_;
   int sample_rate_;
-  rtc::scoped_ptr<int16_t[]> file_;
+  std::unique_ptr<int16_t[]> file_;
   size_t file_pos_;
 };
 
@@ -239,7 +239,7 @@ class FifoAudioStream : public AudioStreamInterface {
   rtc::CriticalSection lock_;
   const size_t frames_per_buffer_;
   const size_t bytes_per_buffer_;
-  rtc::scoped_ptr<AudioBufferList> fifo_;
+  std::unique_ptr<AudioBufferList> fifo_;
   size_t largest_size_;
   size_t total_written_elements_;
   size_t write_count_;
@@ -491,7 +491,7 @@ class MockAudioTransport : public AudioTransport {
   size_t play_count_;
   size_t rec_count_;
   AudioStreamInterface* audio_stream_;
-  rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream_;
+  std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream_;
 };
 
 // AudioDeviceTest test fixture.
@@ -688,11 +688,11 @@ class AudioDeviceTest : public ::testing::Test {
     return volume;
   }
 
-  rtc::scoped_ptr<EventWrapper> test_is_done_;
+  std::unique_ptr<EventWrapper> test_is_done_;
   rtc::scoped_refptr<AudioDeviceModule> audio_device_;
   AudioParameters playout_parameters_;
   AudioParameters record_parameters_;
-  rtc::scoped_ptr<BuildInfo> build_info_;
+  std::unique_ptr<BuildInfo> build_info_;
 };
 
 TEST_F(AudioDeviceTest, ConstructDestruct) {
@@ -935,7 +935,7 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
   NiceMock<MockAudioTransport> mock(kPlayout);
   const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
   std::string file_name = GetFileName(playout_sample_rate());
-  rtc::scoped_ptr<FileAudioStream> file_audio_stream(
+  std::unique_ptr<FileAudioStream> file_audio_stream(
       new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
   mock.HandleCallbacks(test_is_done_.get(),
                        file_audio_stream.get(),
@@ -964,7 +964,7 @@ TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
   EXPECT_EQ(record_channels(), playout_channels());
   EXPECT_EQ(record_sample_rate(), playout_sample_rate());
   NiceMock<MockAudioTransport> mock(kPlayout | kRecording);
-  rtc::scoped_ptr<FifoAudioStream> fifo_audio_stream(
+  std::unique_ptr<FifoAudioStream> fifo_audio_stream(
       new FifoAudioStream(playout_frames_per_10ms_buffer()));
   mock.HandleCallbacks(test_is_done_.get(),
                        fifo_audio_stream.get(),
@@ -994,7 +994,7 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
   EXPECT_EQ(record_channels(), playout_channels());
   EXPECT_EQ(record_sample_rate(), playout_sample_rate());
   NiceMock<MockAudioTransport> mock(kPlayout | kRecording);
-  rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
+  std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
       new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
   mock.HandleCallbacks(test_is_done_.get(),
                        latency_audio_stream.get(),
diff --git a/webrtc/modules/audio_device/android/audio_manager.cc b/webrtc/modules/audio_device/android/audio_manager.cc
index 1d08a6adc0..9174a5b7ab 100644
--- a/webrtc/modules/audio_device/android/audio_manager.cc
+++ b/webrtc/modules/audio_device/android/audio_manager.cc
@@ -16,7 +16,6 @@
 
 #include "webrtc/base/arraysize.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 #include "webrtc/modules/utility/include/helpers_android.h"
 
@@ -32,7 +31,7 @@ namespace webrtc {
 // AudioManager::JavaAudioManager implementation
 AudioManager::JavaAudioManager::JavaAudioManager(
     NativeRegistration* native_reg,
-    rtc::scoped_ptr<GlobalRef> audio_manager)
+    std::unique_ptr<GlobalRef> audio_manager)
     : audio_manager_(std::move(audio_manager)),
       init_(native_reg->GetMethodId("init", "()Z")),
       dispose_(native_reg->GetMethodId("dispose", "()V")),
@@ -67,7 +66,7 @@ bool AudioManager::JavaAudioManager::IsDeviceBlacklistedForOpenSLESUsage() {
 
 // AudioManager implementation
 AudioManager::AudioManager()
-    : j_environment_(JVM::GetInstance()->environment()),
+    : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())),
       audio_layer_(AudioDeviceModule::kPlatformDefaultAudio),
       initialized_(false),
       hardware_aec_(false),
@@ -81,14 +80,14 @@ AudioManager::AudioManager()
       {"nativeCacheAudioParameters",
        "(IIZZZZIIJ)V",
        reinterpret_cast<void*>(&webrtc::AudioManager::CacheAudioParameters)}};
-  j_native_registration_ = j_environment_->RegisterNatives(
+  j_native_registration_ = rtc::ScopedToUnique(j_environment_->RegisterNatives(
       "org/webrtc/voiceengine/WebRtcAudioManager",
-      native_methods, arraysize(native_methods));
+      native_methods, arraysize(native_methods)));
   j_audio_manager_.reset(new JavaAudioManager(
       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)))));
 }
 
 AudioManager::~AudioManager() {
diff --git a/webrtc/modules/audio_device/android/audio_manager.h b/webrtc/modules/audio_device/android/audio_manager.h
index 26caf61afe..b4264a6f8c 100644
--- a/webrtc/modules/audio_device/android/audio_manager.h
+++ b/webrtc/modules/audio_device/android/audio_manager.h
@@ -11,9 +11,10 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_MANAGER_H_
 
+#include <memory>
+
 #include <jni.h>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 #include "webrtc/modules/audio_device/audio_device_config.h"
@@ -39,7 +40,7 @@ class AudioManager {
   class JavaAudioManager {
    public:
     JavaAudioManager(NativeRegistration* native_registration,
-                     rtc::scoped_ptr<GlobalRef> audio_manager);
+                     std::unique_ptr<GlobalRef> audio_manager);
     ~JavaAudioManager();
 
     bool Init();
@@ -48,7 +49,7 @@ class AudioManager {
     bool IsDeviceBlacklistedForOpenSLESUsage();
 
    private:
-    rtc::scoped_ptr<GlobalRef> audio_manager_;
+    std::unique_ptr<GlobalRef> audio_manager_;
     jmethodID init_;
     jmethodID dispose_;
     jmethodID is_communication_mode_enabled_;
@@ -128,13 +129,13 @@ class AudioManager {
   AttachCurrentThreadIfNeeded attach_thread_if_needed_;
 
   // Wraps the JNI interface pointer and methods associated with it.
-  rtc::scoped_ptr<JNIEnvironment> j_environment_;
+  std::unique_ptr<JNIEnvironment> j_environment_;
 
   // Contains factory method for creating the Java object.
-  rtc::scoped_ptr<NativeRegistration> j_native_registration_;
+  std::unique_ptr<NativeRegistration> j_native_registration_;
 
   // Wraps the Java specific parts of the AudioManager.
-  rtc::scoped_ptr<AudioManager::JavaAudioManager> j_audio_manager_;
+  std::unique_ptr<AudioManager::JavaAudioManager> j_audio_manager_;
 
   AudioDeviceModule::AudioLayer audio_layer_;
 
diff --git a/webrtc/modules/audio_device/android/audio_manager_unittest.cc b/webrtc/modules/audio_device/android/audio_manager_unittest.cc
index ddae73067a..d1107e05fd 100644
--- a/webrtc/modules/audio_device/android/audio_manager_unittest.cc
+++ b/webrtc/modules/audio_device/android/audio_manager_unittest.cc
@@ -8,9 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
+
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/format_macros.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/android/build_info.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
 #include "webrtc/modules/audio_device/android/ensure_initialized.h"
@@ -43,7 +44,7 @@ class AudioManagerTest : public ::testing::Test {
     EXPECT_NE(0, audio_manager()->GetDelayEstimateInMilliseconds());
   }
 
-  rtc::scoped_ptr<AudioManager> audio_manager_;
+  std::unique_ptr<AudioManager> audio_manager_;
   AudioParameters playout_parameters_;
   AudioParameters record_parameters_;
 };
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.cc b/webrtc/modules/audio_device/android/audio_record_jni.cc
index 5dda7249ac..5ff59971cf 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.cc
+++ b/webrtc/modules/audio_device/android/audio_record_jni.cc
@@ -31,7 +31,7 @@ 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")),
@@ -74,7 +74,7 @@ bool AudioRecordJni::JavaAudioRecord::EnableBuiltInNS(bool 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),
@@ -93,14 +93,14 @@ AudioRecordJni::AudioRecordJni(AudioManager* audio_manager)
           &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();
diff --git a/webrtc/modules/audio_device/android/audio_record_jni.h b/webrtc/modules/audio_device/android/audio_record_jni.h
index 766316a83a..13194939b6 100644
--- a/webrtc/modules/audio_device/android/audio_record_jni.h
+++ b/webrtc/modules/audio_device/android/audio_record_jni.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_RECORD_JNI_H_
 
+#include <memory>
+
 #include <jni.h>
 
 #include "webrtc/base/thread_checker.h"
@@ -46,7 +48,7 @@ class AudioRecordJni {
   class JavaAudioRecord {
    public:
     JavaAudioRecord(NativeRegistration* native_registration,
-                   rtc::scoped_ptr<GlobalRef> audio_track);
+                   std::unique_ptr<GlobalRef> audio_track);
     ~JavaAudioRecord();
 
     int InitRecording(int sample_rate, size_t channels);
@@ -57,7 +59,7 @@ class AudioRecordJni {
     bool EnableBuiltInNS(bool enable);
 
    private:
-    rtc::scoped_ptr<GlobalRef> audio_record_;
+    std::unique_ptr<GlobalRef> audio_record_;
     jmethodID init_recording_;
     jmethodID start_recording_;
     jmethodID stop_recording_;
@@ -117,13 +119,13 @@ class AudioRecordJni {
   AttachCurrentThreadIfNeeded attach_thread_if_needed_;
 
   // Wraps the JNI interface pointer and methods associated with it.
-  rtc::scoped_ptr<JNIEnvironment> j_environment_;
+  std::unique_ptr<JNIEnvironment> j_environment_;
 
   // Contains factory method for creating the Java object.
-  rtc::scoped_ptr<NativeRegistration> j_native_registration_;
+  std::unique_ptr<NativeRegistration> j_native_registration_;
 
   // Wraps the Java specific parts of the AudioRecordJni class.
-  rtc::scoped_ptr<AudioRecordJni::JavaAudioRecord> j_audio_record_;
+  std::unique_ptr<AudioRecordJni::JavaAudioRecord> j_audio_record_;
 
   // Raw pointer to the audio manger.
   const AudioManager* audio_manager_;
diff --git a/webrtc/modules/audio_device/android/audio_track_jni.cc b/webrtc/modules/audio_device/android/audio_track_jni.cc
index 057e016405..5bf3a5b8d7 100644
--- a/webrtc/modules/audio_device/android/audio_track_jni.cc
+++ b/webrtc/modules/audio_device/android/audio_track_jni.cc
@@ -31,7 +31,7 @@ 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")),
@@ -69,7 +69,7 @@ int AudioTrackJni::JavaAudioTrack::GetStreamVolume() {
 
 // 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),
@@ -86,14 +86,14 @@ AudioTrackJni::AudioTrackJni(AudioManager* audio_manager)
           &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();
diff --git a/webrtc/modules/audio_device/android/audio_track_jni.h b/webrtc/modules/audio_device/android/audio_track_jni.h
index 067dc6c651..32b0b77033 100644
--- a/webrtc/modules/audio_device/android/audio_track_jni.h
+++ b/webrtc/modules/audio_device/android/audio_track_jni.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_AUDIO_TRACK_JNI_H_
 
+#include <memory>
+
 #include <jni.h>
 
 #include "webrtc/base/thread_checker.h"
@@ -42,7 +44,7 @@ class AudioTrackJni {
   class JavaAudioTrack {
    public:
     JavaAudioTrack(NativeRegistration* native_registration,
-                   rtc::scoped_ptr<GlobalRef> audio_track);
+                   std::unique_ptr<GlobalRef> audio_track);
     ~JavaAudioTrack();
 
     void InitPlayout(int sample_rate, int channels);
@@ -53,7 +55,7 @@ class AudioTrackJni {
     int GetStreamVolume();
 
    private:
-    rtc::scoped_ptr<GlobalRef> audio_track_;
+    std::unique_ptr<GlobalRef> audio_track_;
     jmethodID init_playout_;
     jmethodID start_playout_;
     jmethodID stop_playout_;
@@ -113,13 +115,13 @@ class AudioTrackJni {
   AttachCurrentThreadIfNeeded attach_thread_if_needed_;
 
   // Wraps the JNI interface pointer and methods associated with it.
-  rtc::scoped_ptr<JNIEnvironment> j_environment_;
+  std::unique_ptr<JNIEnvironment> j_environment_;
 
   // Contains factory method for creating the Java object.
-  rtc::scoped_ptr<NativeRegistration> j_native_registration_;
+  std::unique_ptr<NativeRegistration> j_native_registration_;
 
   // Wraps the Java specific parts of the AudioTrackJni class.
-  rtc::scoped_ptr<AudioTrackJni::JavaAudioTrack> j_audio_track_;
+  std::unique_ptr<AudioTrackJni::JavaAudioTrack> j_audio_track_;
 
   // Contains audio parameters provided to this class at construction by the
   // AudioManager.
diff --git a/webrtc/modules/audio_device/android/build_info.cc b/webrtc/modules/audio_device/android/build_info.cc
index 6289697073..c6cecc96c5 100644
--- a/webrtc/modules/audio_device/android/build_info.cc
+++ b/webrtc/modules/audio_device/android/build_info.cc
@@ -15,7 +15,7 @@
 namespace webrtc {
 
 BuildInfo::BuildInfo()
-    : j_environment_(JVM::GetInstance()->environment()),
+    : j_environment_(rtc::ScopedToUnique(JVM::GetInstance()->environment())),
       j_build_info_(JVM::GetInstance()->GetClass(
           "org/webrtc/voiceengine/BuildInfo")) {
 }
diff --git a/webrtc/modules/audio_device/android/build_info.h b/webrtc/modules/audio_device/android/build_info.h
index 1490fa0772..4a4c30e836 100644
--- a/webrtc/modules/audio_device/android/build_info.h
+++ b/webrtc/modules/audio_device/android/build_info.h
@@ -12,6 +12,7 @@
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_BUILD_INFO_H_
 
 #include <jni.h>
+#include <memory>
 #include <string>
 
 #include "webrtc/modules/utility/include/jvm_android.h"
@@ -55,7 +56,7 @@ class BuildInfo {
 
   // Provides access to the JNIEnv interface pointer and the JavaToStdString()
   // method which is used to translate Java strings to std strings.
-  rtc::scoped_ptr<JNIEnvironment> j_environment_;
+  std::unique_ptr<JNIEnvironment> j_environment_;
 
   // Holds the jclass object and provides access to CallStaticObjectMethod().
   // Used by GetStringFromJava() during construction only.
diff --git a/webrtc/modules/audio_device/android/opensles_player.h b/webrtc/modules/audio_device/android/opensles_player.h
index fa9e931218..084546dbf7 100644
--- a/webrtc/modules/audio_device/android/opensles_player.h
+++ b/webrtc/modules/audio_device/android/opensles_player.h
@@ -11,11 +11,12 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_ANDROID_OPENSLES_PLAYER_H_
 
+#include <memory>
+
 #include <SLES/OpenSLES.h>
 #include <SLES/OpenSLES_Android.h>
 #include <SLES/OpenSLES_AndroidConfiguration.h>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/android/audio_common.h"
 #include "webrtc/modules/audio_device/android/audio_manager.h"
@@ -150,7 +151,7 @@ class OpenSLESPlayer {
   // Queue of audio buffers to be used by the player object for rendering
   // audio. They will be used in a Round-robin way and the size of each buffer
   // is given by FineAudioBuffer::RequiredBufferSizeBytes().
-  rtc::scoped_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers];
+  std::unique_ptr<SLint8[]> audio_buffers_[kNumOfOpenSLESBuffers];
 
   // FineAudioBuffer takes an AudioDeviceBuffer which delivers audio data
   // in chunks of 10ms. It then allows for this data to be pulled in
@@ -162,7 +163,7 @@ class OpenSLESPlayer {
   // in each callback (one every 5ms). This class can then ask for 240 and the
   // FineAudioBuffer will ask WebRTC for new data only every second callback
   // and also cach non-utilized audio.
-  rtc::scoped_ptr<FineAudioBuffer> fine_buffer_;
+  std::unique_ptr<FineAudioBuffer> fine_buffer_;
 
   // Keeps track of active audio buffer 'n' in the audio_buffers_[n] queue.
   // Example (kNumOfOpenSLESBuffers = 2): counts 0, 1, 0, 1, ...
diff --git a/webrtc/modules/audio_device/audio_device_impl.h b/webrtc/modules/audio_device/audio_device_impl.h
index 5109693964..c7312bf4b0 100644
--- a/webrtc/modules/audio_device/audio_device_impl.h
+++ b/webrtc/modules/audio_device/audio_device_impl.h
@@ -13,8 +13,9 @@
 
 #if defined(WEBRTC_INCLUDE_INTERNAL_AUDIO_DEVICE)
 
+#include <memory>
+
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/audio_device_buffer.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
 
@@ -216,7 +217,7 @@ class AudioDeviceModuleImpl : public AudioDeviceModule {
 
   AudioDeviceBuffer _audioDeviceBuffer;
 #if defined(WEBRTC_ANDROID)
-  rtc::scoped_ptr<AudioManager> _audioManagerAndroid;
+  std::unique_ptr<AudioManager> _audioManagerAndroid;
 #endif
   int32_t _id;
   AudioLayer _platformAudioLayer;
diff --git a/webrtc/modules/audio_device/dummy/file_audio_device.h b/webrtc/modules/audio_device/dummy/file_audio_device.h
index 77179409ea..be6fa3ddee 100644
--- a/webrtc/modules/audio_device/dummy/file_audio_device.h
+++ b/webrtc/modules/audio_device/dummy/file_audio_device.h
@@ -13,6 +13,7 @@
 
 #include <stdio.h>
 
+#include <memory>
 #include <string>
 
 #include "webrtc/modules/audio_device/audio_device_generic.h"
@@ -182,8 +183,8 @@ class FileAudioDevice : public AudioDeviceGeneric {
   size_t _playoutFramesIn10MS;
 
   // TODO(pbos): Make plain members instead of pointers and stop resetting them.
-  rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
-  rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
+  std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
+  std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
 
   bool _playing;
   bool _recording;
diff --git a/webrtc/modules/audio_device/fine_audio_buffer.h b/webrtc/modules/audio_device/fine_audio_buffer.h
index 4ab5cd268c..478e0c6391 100644
--- a/webrtc/modules/audio_device/fine_audio_buffer.h
+++ b/webrtc/modules/audio_device/fine_audio_buffer.h
@@ -11,7 +11,8 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_FINE_AUDIO_BUFFER_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -86,14 +87,14 @@ class FineAudioBuffer {
   // Number of audio bytes per 10ms.
   const size_t bytes_per_10_ms_;
   // Storage for output samples that are not yet asked for.
-  rtc::scoped_ptr<int8_t[]> playout_cache_buffer_;
+  std::unique_ptr<int8_t[]> playout_cache_buffer_;
   // Location of first unread output sample.
   size_t playout_cached_buffer_start_;
   // Number of bytes stored in output (contain samples to be played out) cache.
   size_t playout_cached_bytes_;
   // Storage for input samples that are about to be delivered to the WebRTC
   // ADB or remains from the last successful delivery of a 10ms audio buffer.
-  rtc::scoped_ptr<int8_t[]> record_cache_buffer_;
+  std::unique_ptr<int8_t[]> record_cache_buffer_;
   // Required (max) size in bytes of the |record_cache_buffer_|.
   const size_t required_record_buffer_size_bytes_;
   // Number of bytes in input (contains recorded samples) cache.
diff --git a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
index 6666364c9e..ef189d1fef 100644
--- a/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
+++ b/webrtc/modules/audio_device/fine_audio_buffer_unittest.cc
@@ -15,7 +15,6 @@
 
 #include "testing/gmock/include/gmock/gmock.h"
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_device/mock_audio_device_buffer.h"
 
 using ::testing::_;
@@ -118,9 +117,9 @@ void RunFineBufferTest(int sample_rate, int frame_size_in_samples) {
   FineAudioBuffer fine_buffer(&audio_device_buffer, kFrameSizeBytes,
                               sample_rate);
 
-  rtc::scoped_ptr<int8_t[]> out_buffer;
+  std::unique_ptr<int8_t[]> out_buffer;
   out_buffer.reset(new int8_t[fine_buffer.RequiredPlayoutBufferSizeBytes()]);
-  rtc::scoped_ptr<int8_t[]> in_buffer;
+  std::unique_ptr<int8_t[]> in_buffer;
   in_buffer.reset(new int8_t[kFrameSizeBytes]);
   for (int i = 0; i < kNumberOfFrames; ++i) {
     fine_buffer.GetPlayoutData(out_buffer.get());
diff --git a/webrtc/modules/audio_device/ios/audio_device_ios.h b/webrtc/modules/audio_device/ios/audio_device_ios.h
index c4eb0d6f64..73208864d2 100644
--- a/webrtc/modules/audio_device/ios/audio_device_ios.h
+++ b/webrtc/modules/audio_device/ios/audio_device_ios.h
@@ -11,9 +11,10 @@
 #ifndef WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 #define WEBRTC_MODULES_AUDIO_DEVICE_IOS_AUDIO_DEVICE_IOS_H_
 
+#include <memory>
+
 #include <AudioUnit/AudioUnit.h>
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 
@@ -256,11 +257,11 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
   // can provide audio data frames of size 128 and these are accumulated until
   // enough data to supply one 10ms call exists. This 10ms chunk is then sent
   // to WebRTC and the remaining part is stored.
-  rtc::scoped_ptr<FineAudioBuffer> fine_audio_buffer_;
+  std::unique_ptr<FineAudioBuffer> fine_audio_buffer_;
 
   // Extra audio buffer to be used by the playout side for rendering audio.
   // The buffer size is given by FineAudioBuffer::RequiredBufferSizeBytes().
-  rtc::scoped_ptr<SInt8[]> playout_audio_buffer_;
+  std::unique_ptr<SInt8[]> playout_audio_buffer_;
 
   // Provides a mechanism for encapsulating one or more buffers of audio data.
   // Only used on the recording side.
@@ -268,7 +269,7 @@ class AudioDeviceIOS : public AudioDeviceGeneric {
 
   // Temporary storage for recorded data. AudioUnitRender() renders into this
   // array as soon as a frame of the desired buffer size has been recorded.
-  rtc::scoped_ptr<SInt8[]> record_audio_buffer_;
+  std::unique_ptr<SInt8[]> record_audio_buffer_;
 
   // Set to 1 when recording is active and 0 otherwise.
   volatile int recording_;
diff --git a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
index 076a67430d..4dfb073fa9 100644
--- a/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
+++ b/webrtc/modules/audio_device/ios/audio_device_unittest_ios.cc
@@ -11,6 +11,7 @@
 #include <algorithm>
 #include <limits>
 #include <list>
+#include <memory>
 #include <numeric>
 #include <string>
 #include <vector>
@@ -21,7 +22,6 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/format_macros.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/scoped_ref_ptr.h"
 #include "webrtc/modules/audio_device/audio_device_impl.h"
 #include "webrtc/modules/audio_device/include/audio_device.h"
@@ -145,7 +145,7 @@ class FileAudioStream : public AudioStreamInterface {
  private:
   size_t file_size_in_bytes_;
   int sample_rate_;
-  rtc::scoped_ptr<int16_t[]> file_;
+  std::unique_ptr<int16_t[]> file_;
   size_t file_pos_;
 };
 
@@ -233,7 +233,7 @@ class FifoAudioStream : public AudioStreamInterface {
   rtc::CriticalSection lock_;
   const size_t frames_per_buffer_;
   const size_t bytes_per_buffer_;
-  rtc::scoped_ptr<AudioBufferList> fifo_;
+  std::unique_ptr<AudioBufferList> fifo_;
   size_t largest_size_;
   size_t total_written_elements_;
   size_t write_count_;
@@ -593,7 +593,7 @@ class AudioDeviceTest : public ::testing::Test {
     EXPECT_FALSE(audio_device()->Recording());
   }
 
-  rtc::scoped_ptr<EventWrapper> test_is_done_;
+  std::unique_ptr<EventWrapper> test_is_done_;
   rtc::scoped_refptr<AudioDeviceModule> audio_device_;
   AudioParameters playout_parameters_;
   AudioParameters record_parameters_;
@@ -761,7 +761,7 @@ TEST_F(AudioDeviceTest, RunPlayoutWithFileAsSource) {
   NiceMock<MockAudioTransport> mock(kPlayout);
   const int num_callbacks = kFilePlayTimeInSec * kNumCallbacksPerSecond;
   std::string file_name = GetFileName(playout_sample_rate());
-  rtc::scoped_ptr<FileAudioStream> file_audio_stream(
+  std::unique_ptr<FileAudioStream> file_audio_stream(
       new FileAudioStream(num_callbacks, file_name, playout_sample_rate()));
   mock.HandleCallbacks(test_is_done_.get(), file_audio_stream.get(),
                        num_callbacks);
@@ -795,7 +795,7 @@ TEST_F(AudioDeviceTest, RunPlayoutAndRecordingInFullDuplex) {
   EXPECT_EQ(record_channels(), playout_channels());
   EXPECT_EQ(record_sample_rate(), playout_sample_rate());
   NiceMock<MockAudioTransport> mock(kPlayout | kRecording);
-  rtc::scoped_ptr<FifoAudioStream> fifo_audio_stream(
+  std::unique_ptr<FifoAudioStream> fifo_audio_stream(
       new FifoAudioStream(playout_frames_per_10ms_buffer()));
   mock.HandleCallbacks(test_is_done_.get(), fifo_audio_stream.get(),
                        kFullDuplexTimeInSec * kNumCallbacksPerSecond);
@@ -824,7 +824,7 @@ TEST_F(AudioDeviceTest, DISABLED_MeasureLoopbackLatency) {
   EXPECT_EQ(record_channels(), playout_channels());
   EXPECT_EQ(record_sample_rate(), playout_sample_rate());
   NiceMock<MockAudioTransport> mock(kPlayout | kRecording);
-  rtc::scoped_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
+  std::unique_ptr<LatencyMeasuringAudioStream> latency_audio_stream(
       new LatencyMeasuringAudioStream(playout_frames_per_10ms_buffer()));
   mock.HandleCallbacks(test_is_done_.get(), latency_audio_stream.get(),
                        kMeasureLatencyTimeInSec * kNumCallbacksPerSecond);
diff --git a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
index 4a1a5191be..340e96398f 100644
--- a/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_device_alsa_linux.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_ALSA_LINUX_H
 
+#include <memory>
+
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/linux/audio_mixer_manager_alsa_linux.h"
@@ -187,8 +189,8 @@ private:
 
     // TODO(pbos): Make plain members and start/stop instead of resetting these
     // pointers. A thread can be reused.
-    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
-    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
+    std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
+    std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
 
     int32_t _id;
 
diff --git a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
index de8df0be1b..a0a15e5f0e 100644
--- a/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
+++ b/webrtc/modules/audio_device/linux/audio_device_pulse_linux.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_PULSE_LINUX_H
 
+#include <memory>
+
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
@@ -284,9 +286,9 @@ private:
     EventWrapper& _recStartEvent;
     EventWrapper& _playStartEvent;
 
-    // TODO(pbos): Remove scoped_ptr and use directly without resetting.
-    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadPlay;
-    rtc::scoped_ptr<rtc::PlatformThread> _ptrThreadRec;
+    // TODO(pbos): Remove unique_ptr and use directly without resetting.
+    std::unique_ptr<rtc::PlatformThread> _ptrThreadPlay;
+    std::unique_ptr<rtc::PlatformThread> _ptrThreadRec;
     int32_t _id;
 
     AudioMixerManagerLinuxPulse _mixerManager;
diff --git a/webrtc/modules/audio_device/mac/audio_device_mac.h b/webrtc/modules/audio_device/mac/audio_device_mac.h
index ca3a51997d..102c67e97c 100644
--- a/webrtc/modules/audio_device/mac/audio_device_mac.h
+++ b/webrtc/modules/audio_device/mac/audio_device_mac.h
@@ -11,7 +11,8 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_MAC_H
 
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/mac/audio_mixer_manager_mac.h"
@@ -290,10 +291,10 @@ class AudioDeviceMac : public AudioDeviceGeneric {
   // TODO(pbos): Replace with direct members, just start/stop, no need to
   // recreate the thread.
   // Only valid/running between calls to StartRecording and StopRecording.
-  rtc::scoped_ptr<rtc::PlatformThread> capture_worker_thread_;
+  std::unique_ptr<rtc::PlatformThread> capture_worker_thread_;
 
   // Only valid/running between calls to StartPlayout and StopPlayout.
-  rtc::scoped_ptr<rtc::PlatformThread> render_worker_thread_;
+  std::unique_ptr<rtc::PlatformThread> render_worker_thread_;
 
   int32_t _id;
 
diff --git a/webrtc/modules/audio_device/test/audio_device_test_api.cc b/webrtc/modules/audio_device/test/audio_device_test_api.cc
index f2861ec3e5..a564e35584 100644
--- a/webrtc/modules/audio_device/test/audio_device_test_api.cc
+++ b/webrtc/modules/audio_device/test/audio_device_test_api.cc
@@ -12,6 +12,8 @@
 #include <stdio.h>
 #include <string.h>
 
+#include <memory>
+
 #include "webrtc/modules/audio_device/test/audio_device_test_defines.h"
 
 #include "testing/gtest/include/gtest/gtest.h"
@@ -140,7 +142,8 @@ class AudioDeviceAPITest: public testing::Test {
   virtual ~AudioDeviceAPITest() {}
 
   static void SetUpTestCase() {
-    process_thread_ = ProcessThread::Create("ProcessThread");
+    process_thread_ =
+        rtc::ScopedToUnique(ProcessThread::Create("ProcessThread"));
     process_thread_->Start();
 
     // Windows:
@@ -300,7 +303,7 @@ class AudioDeviceAPITest: public testing::Test {
 
   // TODO(henrika): Get rid of globals.
   static bool linux_alsa_;
-  static rtc::scoped_ptr<ProcessThread> process_thread_;
+  static std::unique_ptr<ProcessThread> process_thread_;
   static AudioDeviceModule* audio_device_;
   static AudioTransportAPI* audio_transport_;
   static AudioEventObserverAPI* event_observer_;
@@ -308,7 +311,7 @@ class AudioDeviceAPITest: public testing::Test {
 
 // Must be initialized like this to handle static SetUpTestCase() above.
 bool AudioDeviceAPITest::linux_alsa_ = false;
-rtc::scoped_ptr<ProcessThread> AudioDeviceAPITest::process_thread_;
+std::unique_ptr<ProcessThread> AudioDeviceAPITest::process_thread_;
 AudioDeviceModule* AudioDeviceAPITest::audio_device_ = NULL;
 AudioTransportAPI* AudioDeviceAPITest::audio_transport_ = NULL;
 AudioEventObserverAPI* AudioDeviceAPITest::event_observer_ = NULL;
diff --git a/webrtc/modules/audio_device/test/func_test_manager.cc b/webrtc/modules/audio_device/test/func_test_manager.cc
index 0a2963e9ae..bb7686c6c1 100644
--- a/webrtc/modules/audio_device/test/func_test_manager.cc
+++ b/webrtc/modules/audio_device/test/func_test_manager.cc
@@ -594,8 +594,8 @@ FuncTestManager::~FuncTestManager()
 
 int32_t FuncTestManager::Init()
 {
-  EXPECT_TRUE((_processThread = ProcessThread::Create("ProcessThread")) !=
-              NULL);
+    EXPECT_TRUE((_processThread = rtc::ScopedToUnique(
+                     ProcessThread::Create("ProcessThread"))) != NULL);
     if (_processThread == NULL)
     {
         return -1;
@@ -832,8 +832,8 @@ int32_t FuncTestManager::TestAudioLayerSelection()
         // ==================================================
         // Next, try to make fresh start with new audio layer
 
-        EXPECT_TRUE((_processThread = ProcessThread::Create("ProcessThread")) !=
-                    NULL);
+        EXPECT_TRUE((_processThread = rtc::ScopedToUnique(
+                         ProcessThread::Create("ProcessThread"))) != NULL);
         if (_processThread == NULL)
         {
             return -1;
diff --git a/webrtc/modules/audio_device/test/func_test_manager.h b/webrtc/modules/audio_device/test/func_test_manager.h
index b7cc81cc1a..2b8a19beb1 100644
--- a/webrtc/modules/audio_device/test/func_test_manager.h
+++ b/webrtc/modules/audio_device/test/func_test_manager.h
@@ -12,6 +12,7 @@
 #define WEBRTC_AUDIO_DEVICE_FUNC_TEST_MANAGER_H
 
 #include <list>
+#include <memory>
 #include <string>
 
 #include "webrtc/common_audio/resampler/include/resampler.h"
@@ -206,7 +207,7 @@ private:
     std::string _playoutFile16;
     std::string _playoutFile8;
 
-    rtc::scoped_ptr<ProcessThread> _processThread;
+    std::unique_ptr<ProcessThread> _processThread;
     AudioDeviceModule* _audioDevice;
     AudioEventObserver* _audioEventObserver;
     AudioTransportImpl* _audioTransport;
diff --git a/webrtc/modules/audio_device/win/audio_device_wave_win.h b/webrtc/modules/audio_device/win/audio_device_wave_win.h
index a1cfc6acbf..402d575d8a 100644
--- a/webrtc/modules/audio_device/win/audio_device_wave_win.h
+++ b/webrtc/modules/audio_device/win/audio_device_wave_win.h
@@ -11,6 +11,8 @@
 #ifndef WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
 #define WEBRTC_AUDIO_DEVICE_AUDIO_DEVICE_WAVE_WIN_H
 
+#include <memory>
+
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/audio_device/audio_device_generic.h"
 #include "webrtc/modules/audio_device/win/audio_mixer_manager_win.h"
@@ -222,8 +224,8 @@ private:
     HANDLE                                  _hShutdownSetVolumeEvent;
     HANDLE                                  _hSetCaptureVolumeEvent;
 
-    // TODO(pbos): Remove scoped_ptr usage and use PlatformThread directly
-    rtc::scoped_ptr<rtc::PlatformThread>    _ptrThread;
+    // TODO(pbos): Remove unique_ptr usage and use PlatformThread directly
+    std::unique_ptr<rtc::PlatformThread>    _ptrThread;
 
     CriticalSectionWrapper&                 _critSectCb;