Replace scoped_ptr with unique_ptr in webrtc/modules/audio_coding/
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1694073002
Cr-Commit-Position: refs/heads/master@{#11622}
diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
index 855a39e..ec3f4fc 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.cc
@@ -13,6 +13,8 @@
#include <assert.h>
#include <stdio.h>
+#include <memory>
+
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/audio_sink.h"
@@ -151,7 +153,7 @@
}
void AcmReceiveTestOldApi::Run() {
- for (rtc::scoped_ptr<Packet> packet(packet_source_->NextPacket()); packet;
+ for (std::unique_ptr<Packet> packet(packet_source_->NextPacket()); packet;
packet.reset(packet_source_->NextPacket())) {
// Pull audio until time to insert packet.
while (clock_.TimeInMilliseconds() < packet->time_ms()) {
diff --git a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
index 3010ec7..6952641 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h
@@ -11,10 +11,10 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVE_TEST_OLDAPI_H_
+#include <memory>
#include <string>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/include/clock.h"
namespace webrtc {
@@ -61,7 +61,7 @@
virtual void AfterGetAudio() {}
SimulatedClock clock_;
- rtc::scoped_ptr<AudioCodingModule> acm_;
+ std::unique_ptr<AudioCodingModule> acm_;
PacketSource* packet_source_;
AudioSink* audio_sink_;
int output_freq_hz_;
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver.h b/webrtc/modules/audio_coding/acm2/acm_receiver.h
index d1ca504..f5ceb61 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver.h
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver.h
@@ -12,13 +12,13 @@
#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_RECEIVER_H_
#include <map>
+#include <memory>
#include <string>
#include <vector>
#include "webrtc/base/array_view.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/optional.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_audio/vad/include/webrtc_vad.h"
#include "webrtc/engine_configurations.h"
@@ -288,8 +288,8 @@
ACMResampler resampler_ GUARDED_BY(crit_sect_);
// Used in GetAudio, declared as member to avoid allocating every 10ms.
// TODO(henrik.lundin) Stack-allocate in GetAudio instead?
- rtc::scoped_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
- rtc::scoped_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
+ std::unique_ptr<int16_t[]> audio_buffer_ GUARDED_BY(crit_sect_);
+ std::unique_ptr<int16_t[]> last_audio_buffer_ GUARDED_BY(crit_sect_);
CallStatistics call_stats_ GUARDED_BY(crit_sect_);
NetEq* neteq_;
// Decoders map is keyed by payload type
diff --git a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
index 24ecc69..a0f4e0e 100644
--- a/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/acm_receiver_unittest_oldapi.cc
@@ -11,9 +11,9 @@
#include "webrtc/modules/audio_coding/acm2/acm_receiver.h"
#include <algorithm> // std::min
+#include <memory>
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h"
#include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
@@ -153,9 +153,9 @@
return 0;
}
- rtc::scoped_ptr<AcmReceiver> receiver_;
+ std::unique_ptr<AcmReceiver> receiver_;
rtc::ArrayView<const CodecInst> codecs_;
- rtc::scoped_ptr<AudioCodingModule> acm_;
+ std::unique_ptr<AudioCodingModule> acm_;
WebRtcRTPHeader rtp_header_;
uint32_t timestamp_;
bool packet_sent_; // Set when SendData is called reset when inserting audio.
diff --git a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
index ce68196..cfee353 100644
--- a/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
+++ b/webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h
@@ -11,10 +11,10 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
#define WEBRTC_MODULES_AUDIO_CODING_ACM2_ACM_SEND_TEST_OLDAPI_H_
+#include <memory>
#include <vector>
#include "webrtc/base/constructormagic.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
#include "webrtc/modules/audio_coding/neteq/tools/packet_source.h"
#include "webrtc/system_wrappers/include/clock.h"
@@ -68,7 +68,7 @@
Packet* CreatePacket();
SimulatedClock clock_;
- rtc::scoped_ptr<AudioCodingModule> acm_;
+ std::unique_ptr<AudioCodingModule> acm_;
InputAudioFile* audio_source_;
int source_rate_hz_;
const size_t input_block_size_samples_;
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
index 102396f..cdf4944 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_impl.h
@@ -11,12 +11,12 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
#define WEBRTC_MODULES_AUDIO_CODING_ACM2_AUDIO_CODING_MODULE_IMPL_H_
+#include <memory>
#include <string>
#include <vector>
#include "webrtc/base/buffer.h"
#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/common_types.h"
#include "webrtc/engine_configurations.h"
@@ -253,7 +253,7 @@
CodecManager codec_manager;
RentACodec rent_a_codec;
};
- rtc::scoped_ptr<EncoderFactory> encoder_factory_ GUARDED_BY(acm_crit_sect_);
+ std::unique_ptr<EncoderFactory> encoder_factory_ GUARDED_BY(acm_crit_sect_);
// Current encoder stack, either obtained from
// encoder_factory_->rent_a_codec.RentEncoderStack or provided by a call to
@@ -269,7 +269,7 @@
// IMPORTANT: this variable is only used in IncomingPayload(), therefore,
// no lock acquired when interacting with this variable. If it is going to
// be used in other methods, locks need to be taken.
- rtc::scoped_ptr<WebRtcRTPHeader> aux_rtp_header_;
+ std::unique_ptr<WebRtcRTPHeader> aux_rtp_header_;
bool receiver_initialized_ GUARDED_BY(acm_crit_sect_);
diff --git a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
index aba96e4..f169d05 100644
--- a/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
+++ b/webrtc/modules/audio_coding/acm2/audio_coding_module_unittest_oldapi.cc
@@ -10,13 +10,13 @@
#include <stdio.h>
#include <string.h>
+#include <memory>
#include <vector>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/md5digest.h"
#include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_annotations.h"
#include "webrtc/modules/audio_coding/acm2/acm_receive_test_oldapi.h"
#include "webrtc/modules/audio_coding/acm2/acm_send_test_oldapi.h"
@@ -225,8 +225,8 @@
}
const int id_;
- rtc::scoped_ptr<RtpUtility> rtp_utility_;
- rtc::scoped_ptr<AudioCodingModule> acm_;
+ std::unique_ptr<RtpUtility> rtp_utility_;
+ std::unique_ptr<AudioCodingModule> acm_;
PacketizationCallbackStubOldApi packet_cb_;
WebRtcRTPHeader rtp_header_;
AudioFrame input_frame_;
@@ -575,13 +575,13 @@
rtc::PlatformThread send_thread_;
rtc::PlatformThread insert_packet_thread_;
rtc::PlatformThread pull_audio_thread_;
- const rtc::scoped_ptr<EventWrapper> test_complete_;
+ const std::unique_ptr<EventWrapper> test_complete_;
int send_count_;
int insert_packet_count_;
int pull_audio_count_ GUARDED_BY(crit_sect_);
rtc::CriticalSection crit_sect_;
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
- rtc::scoped_ptr<SimulatedClock> fake_clock_;
+ std::unique_ptr<SimulatedClock> fake_clock_;
};
#if defined(WEBRTC_IOS)
@@ -775,7 +775,7 @@
bool CbReceiveImpl() {
SleepMs(1);
const size_t max_encoded_bytes = isac_encoder_->MaxEncodedBytes();
- rtc::scoped_ptr<uint8_t[]> encoded(new uint8_t[max_encoded_bytes]);
+ std::unique_ptr<uint8_t[]> encoded(new uint8_t[max_encoded_bytes]);
AudioEncoder::EncodedInfo info;
{
rtc::CritScope lock(&crit_sect_);
@@ -841,13 +841,13 @@
rtc::PlatformThread receive_thread_;
rtc::PlatformThread codec_registration_thread_;
- const rtc::scoped_ptr<EventWrapper> test_complete_;
+ const std::unique_ptr<EventWrapper> test_complete_;
rtc::CriticalSection crit_sect_;
bool codec_registered_ GUARDED_BY(crit_sect_);
int receive_packet_count_ GUARDED_BY(crit_sect_);
int64_t next_insert_packet_time_ms_ GUARDED_BY(crit_sect_);
- rtc::scoped_ptr<AudioEncoderIsac> isac_encoder_;
- rtc::scoped_ptr<SimulatedClock> fake_clock_;
+ std::unique_ptr<AudioEncoderIsac> isac_encoder_;
+ std::unique_ptr<SimulatedClock> fake_clock_;
test::AudioLoop audio_loop_;
};
@@ -897,7 +897,7 @@
const std::vector<ExternalDecoder>& external_decoders) {
const std::string input_file_name =
webrtc::test::ResourcePath("audio_coding/neteq_universal_new", "rtp");
- rtc::scoped_ptr<test::RtpFileSource> packet_source(
+ std::unique_ptr<test::RtpFileSource> packet_source(
test::RtpFileSource::Create(input_file_name));
#ifdef WEBRTC_ANDROID
// Filter out iLBC and iSAC-swb since they are not supported on Android.
@@ -1199,8 +1199,8 @@
RegisterExternalSendCodec(external_speech_encoder, payload_type));
}
- rtc::scoped_ptr<test::AcmSendTestOldApi> send_test_;
- rtc::scoped_ptr<test::InputAudioFile> audio_source_;
+ std::unique_ptr<test::AcmSendTestOldApi> send_test_;
+ std::unique_ptr<test::InputAudioFile> audio_source_;
uint32_t frame_size_rtp_timestamps_;
int packet_count_;
uint8_t payload_type_;
@@ -1490,8 +1490,8 @@
codec_frame_size_rtp_timestamps));
}
- rtc::scoped_ptr<test::AcmSendTestOldApi> send_test_;
- rtc::scoped_ptr<test::InputAudioFile> audio_source_;
+ std::unique_ptr<test::AcmSendTestOldApi> send_test_;
+ std::unique_ptr<test::InputAudioFile> audio_source_;
};
TEST_F(AcmSetBitRateOldApi, Opus_48khz_20ms_10kbps) {
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager.h b/webrtc/modules/audio_coding/acm2/codec_manager.h
index 9227e13..660a9c0 100644
--- a/webrtc/modules/audio_coding/acm2/codec_manager.h
+++ b/webrtc/modules/audio_coding/acm2/codec_manager.h
@@ -15,7 +15,6 @@
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/thread_checker.h"
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
diff --git a/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc b/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
index dce8f38..8320614 100644
--- a/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/codec_manager_unittest.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <memory>
+
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/codecs/mock/mock_audio_encoder.h"
#include "webrtc/modules/audio_coding/acm2/codec_manager.h"
@@ -21,8 +23,8 @@
namespace {
// Create a MockAudioEncoder with some reasonable default behavior.
-rtc::scoped_ptr<MockAudioEncoder> CreateMockEncoder() {
- auto enc = rtc_make_scoped_ptr(new MockAudioEncoder);
+std::unique_ptr<MockAudioEncoder> CreateMockEncoder() {
+ auto enc = std::unique_ptr<MockAudioEncoder>(new MockAudioEncoder);
EXPECT_CALL(*enc, SampleRateHz()).WillRepeatedly(Return(8000));
EXPECT_CALL(*enc, NumChannels()).WillRepeatedly(Return(1));
EXPECT_CALL(*enc, Max10MsFramesInAPacket()).WillRepeatedly(Return(1));
diff --git a/webrtc/modules/audio_coding/acm2/initial_delay_manager.h b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
index 32dd126..a73141a 100644
--- a/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
+++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager.h
@@ -11,7 +11,6 @@
#ifndef WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
#define WEBRTC_MODULES_AUDIO_CODING_ACM2_INITIAL_DELAY_MANAGER_H_
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/include/module_common_types.h"
namespace webrtc {
diff --git a/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc b/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
index d86d221..bf2850d 100644
--- a/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
+++ b/webrtc/modules/audio_coding/acm2/initial_delay_manager_unittest.cc
@@ -10,6 +10,8 @@
#include <string.h>
+#include <memory>
+
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/modules/audio_coding/acm2/initial_delay_manager.h"
@@ -78,7 +80,7 @@
NextRtpHeader(rtp_info, rtp_receive_timestamp);
}
- rtc::scoped_ptr<InitialDelayManager> manager_;
+ std::unique_ptr<InitialDelayManager> manager_;
WebRtcRTPHeader rtp_info_;
uint32_t rtp_receive_timestamp_;
};
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
index 44a38bb..91c5e4d 100644
--- a/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.cc
@@ -10,6 +10,7 @@
#include "webrtc/modules/audio_coding/acm2/rent_a_codec.h"
+#include <memory>
#include <utility>
#include "webrtc/base/logging.h"
@@ -144,52 +145,53 @@
// Returns a new speech encoder, or null on error.
// TODO(kwiberg): Don't handle errors here (bug 5033)
-rtc::scoped_ptr<AudioEncoder> CreateEncoder(
- const CodecInst& speech_inst,
- LockedIsacBandwidthInfo* bwinfo) {
+std::unique_ptr<AudioEncoder> CreateEncoder(const CodecInst& speech_inst,
+ LockedIsacBandwidthInfo* bwinfo) {
#if defined(WEBRTC_CODEC_ISACFX)
if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
- return rtc_make_scoped_ptr(new AudioEncoderIsacFix(speech_inst, bwinfo));
+ return std::unique_ptr<AudioEncoder>(
+ new AudioEncoderIsacFix(speech_inst, bwinfo));
#endif
#if defined(WEBRTC_CODEC_ISAC)
if (STR_CASE_CMP(speech_inst.plname, "isac") == 0)
- return rtc_make_scoped_ptr(new AudioEncoderIsac(speech_inst, bwinfo));
+ return std::unique_ptr<AudioEncoder>(
+ new AudioEncoderIsac(speech_inst, bwinfo));
#endif
#ifdef WEBRTC_CODEC_OPUS
if (STR_CASE_CMP(speech_inst.plname, "opus") == 0)
- return rtc_make_scoped_ptr(new AudioEncoderOpus(speech_inst));
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderOpus(speech_inst));
#endif
if (STR_CASE_CMP(speech_inst.plname, "pcmu") == 0)
- return rtc_make_scoped_ptr(new AudioEncoderPcmU(speech_inst));
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmU(speech_inst));
if (STR_CASE_CMP(speech_inst.plname, "pcma") == 0)
- return rtc_make_scoped_ptr(new AudioEncoderPcmA(speech_inst));
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcmA(speech_inst));
if (STR_CASE_CMP(speech_inst.plname, "l16") == 0)
- return rtc_make_scoped_ptr(new AudioEncoderPcm16B(speech_inst));
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderPcm16B(speech_inst));
#ifdef WEBRTC_CODEC_ILBC
if (STR_CASE_CMP(speech_inst.plname, "ilbc") == 0)
- return rtc_make_scoped_ptr(new AudioEncoderIlbc(speech_inst));
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderIlbc(speech_inst));
#endif
#ifdef WEBRTC_CODEC_G722
if (STR_CASE_CMP(speech_inst.plname, "g722") == 0)
- return rtc_make_scoped_ptr(new AudioEncoderG722(speech_inst));
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderG722(speech_inst));
#endif
LOG_F(LS_ERROR) << "Could not create encoder of type " << speech_inst.plname;
- return rtc::scoped_ptr<AudioEncoder>();
+ return std::unique_ptr<AudioEncoder>();
}
-rtc::scoped_ptr<AudioEncoder> CreateRedEncoder(AudioEncoder* encoder,
+std::unique_ptr<AudioEncoder> CreateRedEncoder(AudioEncoder* encoder,
int red_payload_type) {
#ifdef WEBRTC_CODEC_RED
AudioEncoderCopyRed::Config config;
config.payload_type = red_payload_type;
config.speech_encoder = encoder;
- return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCopyRed(config));
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderCopyRed(config));
#else
- return rtc::scoped_ptr<AudioEncoder>();
+ return std::unique_ptr<AudioEncoder>();
#endif
}
-rtc::scoped_ptr<AudioEncoder> CreateCngEncoder(AudioEncoder* encoder,
+std::unique_ptr<AudioEncoder> CreateCngEncoder(AudioEncoder* encoder,
int payload_type,
ACMVADMode vad_mode) {
AudioEncoderCng::Config config;
@@ -212,18 +214,18 @@
default:
FATAL();
}
- return rtc::scoped_ptr<AudioEncoder>(new AudioEncoderCng(config));
+ return std::unique_ptr<AudioEncoder>(new AudioEncoderCng(config));
}
-rtc::scoped_ptr<AudioDecoder> CreateIsacDecoder(
+std::unique_ptr<AudioDecoder> CreateIsacDecoder(
LockedIsacBandwidthInfo* bwinfo) {
#if defined(WEBRTC_CODEC_ISACFX)
- return rtc_make_scoped_ptr(new AudioDecoderIsacFix(bwinfo));
+ return std::unique_ptr<AudioDecoder>(new AudioDecoderIsacFix(bwinfo));
#elif defined(WEBRTC_CODEC_ISAC)
- return rtc_make_scoped_ptr(new AudioDecoderIsac(bwinfo));
+ return std::unique_ptr<AudioDecoder>(new AudioDecoderIsac(bwinfo));
#else
FATAL() << "iSAC is not supported.";
- return rtc::scoped_ptr<AudioDecoder>();
+ return std::unique_ptr<AudioDecoder>();
#endif
}
@@ -233,7 +235,7 @@
RentACodec::~RentACodec() = default;
AudioEncoder* RentACodec::RentEncoder(const CodecInst& codec_inst) {
- rtc::scoped_ptr<AudioEncoder> enc =
+ std::unique_ptr<AudioEncoder> enc =
CreateEncoder(codec_inst, &isac_bandwidth_info_);
if (!enc)
return nullptr;
diff --git a/webrtc/modules/audio_coding/acm2/rent_a_codec.h b/webrtc/modules/audio_coding/acm2/rent_a_codec.h
index 3f914ea..dd2dece 100644
--- a/webrtc/modules/audio_coding/acm2/rent_a_codec.h
+++ b/webrtc/modules/audio_coding/acm2/rent_a_codec.h
@@ -13,11 +13,11 @@
#include <stddef.h>
#include <map>
+#include <memory>
#include "webrtc/base/array_view.h"
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/optional.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_coding/codecs/audio_decoder.h"
#include "webrtc/modules/audio_coding/codecs/audio_encoder.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module_typedefs.h"
@@ -229,10 +229,10 @@
AudioDecoder* RentIsacDecoder();
private:
- rtc::scoped_ptr<AudioEncoder> speech_encoder_;
- rtc::scoped_ptr<AudioEncoder> cng_encoder_;
- rtc::scoped_ptr<AudioEncoder> red_encoder_;
- rtc::scoped_ptr<AudioDecoder> isac_decoder_;
+ std::unique_ptr<AudioEncoder> speech_encoder_;
+ std::unique_ptr<AudioEncoder> cng_encoder_;
+ std::unique_ptr<AudioEncoder> red_encoder_;
+ std::unique_ptr<AudioDecoder> isac_decoder_;
LockedIsacBandwidthInfo isac_bandwidth_info_;
RTC_DISALLOW_COPY_AND_ASSIGN(RentACodec);