diff --git a/webrtc/common_audio/vad/include/vad.h b/webrtc/common_audio/vad/include/vad.h
index 087970f58e..86e6f66e0d 100644
--- a/webrtc/common_audio/vad/include/vad.h
+++ b/webrtc/common_audio/vad/include/vad.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_COMMON_AUDIO_VAD_INCLUDE_VAD_H_
 #define WEBRTC_COMMON_AUDIO_VAD_INCLUDE_VAD_H_
 
+#include <memory>
+
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/vad/include/webrtc_vad.h"
 #include "webrtc/typedefs.h"
 
@@ -43,7 +44,7 @@ class Vad {
 };
 
 // Returns a Vad instance that's implemented on top of WebRtcVad.
-rtc::scoped_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness);
+std::unique_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness);
 
 }  // namespace webrtc
 
diff --git a/webrtc/common_audio/vad/vad.cc b/webrtc/common_audio/vad/vad.cc
index 95a162fb92..99d6ffeee6 100644
--- a/webrtc/common_audio/vad/vad.cc
+++ b/webrtc/common_audio/vad/vad.cc
@@ -56,8 +56,8 @@ class VadImpl final : public Vad {
 
 }  // namespace
 
-rtc::scoped_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness) {
-  return rtc::scoped_ptr<Vad>(new VadImpl(aggressiveness));
+std::unique_ptr<Vad> CreateVad(Vad::Aggressiveness aggressiveness) {
+  return std::unique_ptr<Vad>(new VadImpl(aggressiveness));
 }
 
 }  // namespace webrtc
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
index 180166c40c..464655c511 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.cc
@@ -11,6 +11,7 @@
 #include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 
 #include <algorithm>
+#include <memory>
 #include <limits>
 
 namespace webrtc {
@@ -19,12 +20,13 @@ namespace {
 
 const int kMaxFrameSizeMs = 60;
 
-rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> CreateCngInst(
+std::unique_ptr<CNG_enc_inst, CngInstDeleter> CreateCngInst(
     int sample_rate_hz,
     int sid_frame_interval_ms,
     int num_cng_coefficients) {
-  rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst;
-  RTC_CHECK_EQ(0, WebRtcCng_CreateEnc(cng_inst.accept()));
+  CNG_enc_inst* ci;
+  RTC_CHECK_EQ(0, WebRtcCng_CreateEnc(&ci));
+  std::unique_ptr<CNG_enc_inst, CngInstDeleter> cng_inst(ci);
   RTC_CHECK_EQ(0,
                WebRtcCng_InitEnc(cng_inst.get(), sample_rate_hz,
                                  sid_frame_interval_ms, num_cng_coefficients));
@@ -55,7 +57,7 @@ AudioEncoderCng::AudioEncoderCng(const Config& config)
       num_cng_coefficients_(config.num_cng_coefficients),
       sid_frame_interval_ms_(config.sid_frame_interval_ms),
       last_frame_active_(true),
-      vad_(config.vad ? rtc_make_scoped_ptr(config.vad)
+      vad_(config.vad ? std::unique_ptr<Vad>(config.vad)
                       : CreateVad(config.vad_mode)) {
   RTC_CHECK(config.IsOk()) << "Invalid configuration.";
   cng_inst_ = CreateCngInst(SampleRateHz(), sid_frame_interval_ms_,
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
index 87383e2ac5..8a17ac9822 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h
@@ -11,16 +11,17 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_CNG_AUDIO_ENCODER_CNG_H_
 
+#include <memory>
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/common_audio/vad/include/vad.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/cng/webrtc_cng.h"
 
 namespace webrtc {
 
-// Deleter for use with scoped_ptr.
+// Deleter for use with unique_ptr.
 struct CngInstDeleter {
   void operator()(CNG_enc_inst* ptr) const { WebRtcCng_FreeEnc(ptr); }
 };
@@ -84,8 +85,8 @@ class AudioEncoderCng final : public AudioEncoder {
   std::vector<int16_t> speech_buffer_;
   std::vector<uint32_t> rtp_timestamps_;
   bool last_frame_active_;
-  rtc::scoped_ptr<Vad> vad_;
-  rtc::scoped_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
+  std::unique_ptr<Vad> vad_;
+  std::unique_ptr<CNG_enc_inst, CngInstDeleter> cng_inst_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderCng);
 };
diff --git a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
index feb3ed1f0a..fbfd999d39 100644
--- a/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng_unittest.cc
@@ -8,10 +8,10 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_audio/vad/mock/mock_vad.h"
 #include "webrtc/modules/audio_coding/codecs/cng/audio_encoder_cng.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
@@ -185,7 +185,7 @@ class AudioEncoderCngTest : public ::testing::Test {
   }
 
   AudioEncoderCng::Config config_;
-  rtc::scoped_ptr<AudioEncoderCng> cng_;
+  std::unique_ptr<AudioEncoderCng> cng_;
   MockAudioEncoder mock_encoder_;
   MockVad* mock_vad_;  // Ownership is transferred to |cng_|.
   uint32_t timestamp_;
diff --git a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
index b839488628..415511705a 100644
--- a/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
+++ b/webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h
@@ -13,7 +13,7 @@
 
 #include <vector>
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
index 07d767e778..d611226d75 100644
--- a/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
+++ b/webrtc/modules/audio_coding/codecs/g722/audio_encoder_g722.h
@@ -11,8 +11,9 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_G722_AUDIO_ENCODER_G722_H_
 
+#include <memory>
+
 #include "webrtc/base/buffer.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/g722/g722_interface.h"
 
@@ -51,7 +52,7 @@ class AudioEncoderG722 final : public AudioEncoder {
   // The encoder state for one channel.
   struct EncoderState {
     G722EncInst* encoder;
-    rtc::scoped_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
+    std::unique_ptr<int16_t[]> speech_buffer;   // Queued up for encoding.
     rtc::Buffer encoded_buffer;                 // Already encoded.
     EncoderState();
     ~EncoderState();
@@ -64,7 +65,7 @@ class AudioEncoderG722 final : public AudioEncoder {
   const size_t num_10ms_frames_per_packet_;
   size_t num_10ms_frames_buffered_;
   uint32_t first_timestamp_in_buffer_;
-  const rtc::scoped_ptr<EncoderState[]> encoders_;
+  const std::unique_ptr<EncoderState[]> encoders_;
   rtc::Buffer interleave_buffer_;
   RTC_DISALLOW_COPY_AND_ASSIGN(AudioEncoderG722);
 };
diff --git a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
index 102a274642..7b6c075f0a 100644
--- a/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
+++ b/webrtc/modules/audio_coding/codecs/ilbc/audio_encoder_ilbc.h
@@ -11,7 +11,7 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ILBC_AUDIO_ENCODER_ILBC_H_
 
-#include "webrtc/base/scoped_ptr.h"
+#include "webrtc/base/constructormagic.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 #include "webrtc/modules/audio_coding/codecs/ilbc/ilbc.h"
 
diff --git a/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h b/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
index 002af8c392..6bb32c99ed 100644
--- a/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
+++ b/webrtc/modules/audio_coding/codecs/isac/locked_bandwidth_info.h
@@ -12,7 +12,6 @@
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_ISAC_LOCKED_BANDWIDTH_INFO_H_
 
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/modules/audio_coding/codecs/isac/bandwidth_info.h"
 
diff --git a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
index 441e807b4f..0adaeee7f4 100644
--- a/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus_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/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/common_types.h"
 #include "webrtc/modules/audio_coding/codecs/opus/audio_encoder_opus.h"
 
@@ -31,7 +32,7 @@ class AudioEncoderOpusTest : public ::testing::Test {
   }
 
   CodecInst codec_inst_ = kOpusSettings;
-  rtc::scoped_ptr<AudioEncoderOpus> encoder_;
+  std::unique_ptr<AudioEncoderOpus> encoder_;
 };
 
 TEST_F(AudioEncoderOpusTest, DefaultApplicationModeMono) {
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
index 4f9f7ff7bb..560c978b30 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_fec_test.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_coding/codecs/opus/opus_interface.h"
 #include "webrtc/test/testsupport/fileutils.h"
 
@@ -61,9 +62,9 @@ class OpusFecTest : public TestWithParam<coding_param> {
 
   string in_filename_;
 
-  rtc::scoped_ptr<int16_t[]> in_data_;
-  rtc::scoped_ptr<int16_t[]> out_data_;
-  rtc::scoped_ptr<uint8_t[]> bit_stream_;
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
 };
 
 void OpusFecTest::SetUp() {
diff --git a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
index b6112d1881..cdfa62b752 100644
--- a/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/opus/opus_unittest.cc
@@ -7,6 +7,8 @@
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
+
+#include <memory>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
@@ -636,7 +638,7 @@ TEST_P(OpusTest, OpusDecodeRepacketized) {
 
   // Encode & decode.
   int16_t audio_type;
-  rtc::scoped_ptr<int16_t[]> output_data_decode(
+  std::unique_ptr<int16_t[]> output_data_decode(
       new int16_t[kPackets * kOpus20msFrameSamples * channels_]);
   OpusRepacketizer* rp = opus_repacketizer_create();
 
diff --git a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
index 68ca2da77e..34a780b49d 100644
--- a/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
+++ b/webrtc/modules/audio_coding/codecs/pcm16b/audio_encoder_pcm16b.h
@@ -11,7 +11,6 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_PCM16B_AUDIO_ENCODER_PCM16B_H_
 
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/g711/audio_encoder_pcm.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
index 2f53765389..4f31ecf0fc 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h
@@ -14,7 +14,6 @@
 #include <vector>
 
 #include "webrtc/base/buffer.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
 
 namespace webrtc {
diff --git a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
index 22601b6597..a194fb98aa 100644
--- a/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
+++ b/webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red_unittest.cc
@@ -8,11 +8,11 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
+#include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/modules/audio_coding/codecs/red/audio_encoder_copy_red.h"
 #include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
 
@@ -68,7 +68,7 @@ class AudioEncoderCopyRedTest : public ::testing::Test {
   }
 
   MockAudioEncoder mock_encoder_;
-  rtc::scoped_ptr<AudioEncoderCopyRed> red_;
+  std::unique_ptr<AudioEncoderCopyRed> red_;
   uint32_t timestamp_;
   int16_t audio_[kMaxNumSamples];
   const int sample_rate_hz_;
diff --git a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
index fb7b3e5b1e..3560eac342 100644
--- a/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
+++ b/webrtc/modules/audio_coding/codecs/tools/audio_codec_speed_test.h
@@ -11,9 +11,9 @@
 #ifndef WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
 #define WEBRTC_MODULES_AUDIO_CODING_CODECS_TOOLS_AUDIO_CODEC_SPEED_TEST_H_
 
+#include <memory>
 #include <string>
 #include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/typedefs.h"
 
 namespace webrtc {
@@ -61,11 +61,11 @@ class AudioCodecSpeedTest : public testing::TestWithParam<coding_param> {
   // Expected output number of samples-per-channel in a frame.
   size_t output_length_sample_;
 
-  rtc::scoped_ptr<int16_t[]> in_data_;
-  rtc::scoped_ptr<int16_t[]> out_data_;
+  std::unique_ptr<int16_t[]> in_data_;
+  std::unique_ptr<int16_t[]> out_data_;
   size_t data_pointer_;
   size_t loop_length_samples_;
-  rtc::scoped_ptr<uint8_t[]> bit_stream_;
+  std::unique_ptr<uint8_t[]> bit_stream_;
 
   // Maximum number of bytes in output bitstream for a frame of audio.
   size_t max_bytes_;