Replace scoped_ptr with unique_ptr in webrtc/voice_engine/
Also introduce a pair of scoped_ptr <-> unique_ptr conversion
functions. By using them judiciously, we can keep these CL:s small and
avoid having to convert enormous amounts of code at once.
BUG=webrtc:5520
Review URL: https://codereview.webrtc.org/1702983002
Cr-Original-Commit-Position: refs/heads/master@{#11658}
Cr-Mirrored-From: https://chromium.googlesource.com/external/webrtc
Cr-Mirrored-Commit: b7f89d6e668602e46a20e3a9927790ff34029b2a
diff --git a/voice_engine/channel.cc b/voice_engine/channel.cc
index c078d20..9e27ce8 100644
--- a/voice_engine/channel.cc
+++ b/voice_engine/channel.cc
@@ -1071,7 +1071,7 @@
return 0;
}
-void Channel::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void Channel::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
rtc::CritScope cs(&_callbackCritSect);
audio_sink_ = std::move(sink);
}
@@ -3265,7 +3265,7 @@
// TODO(andrew): refactor Mix functions here and in transmit_mixer.cc to use
// a shared helper.
int32_t Channel::MixOrReplaceAudioWithFile(int mixingFrequency) {
- rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
+ std::unique_ptr<int16_t[]> fileBuffer(new int16_t[640]);
size_t fileSamples(0);
{
@@ -3313,7 +3313,7 @@
int32_t Channel::MixAudioWithFile(AudioFrame& audioFrame, int mixingFrequency) {
assert(mixingFrequency <= 48000);
- rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[960]);
+ std::unique_ptr<int16_t[]> fileBuffer(new int16_t[960]);
size_t fileSamples(0);
{
diff --git a/voice_engine/channel.h b/voice_engine/channel.h
index 60c751e..0e87252 100644
--- a/voice_engine/channel.h
+++ b/voice_engine/channel.h
@@ -11,9 +11,10 @@
#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_H_
#define WEBRTC_VOICE_ENGINE_CHANNEL_H_
+#include <memory>
+
#include "webrtc/audio/audio_sink.h"
#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_coding/include/audio_coding_module.h"
@@ -193,7 +194,7 @@
rtc::CriticalSection* callbackCritSect);
int32_t UpdateLocalTimeStamp();
- void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+ void SetSink(std::unique_ptr<AudioSinkInterface> sink);
// API methods
@@ -493,15 +494,15 @@
RtcEventLog* const event_log_;
- rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
- rtc::scoped_ptr<RTPPayloadRegistry> rtp_payload_registry_;
- rtc::scoped_ptr<ReceiveStatistics> rtp_receive_statistics_;
- rtc::scoped_ptr<StatisticsProxy> statistics_proxy_;
- rtc::scoped_ptr<RtpReceiver> rtp_receiver_;
+ std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
+ std::unique_ptr<RTPPayloadRegistry> rtp_payload_registry_;
+ std::unique_ptr<ReceiveStatistics> rtp_receive_statistics_;
+ std::unique_ptr<StatisticsProxy> statistics_proxy_;
+ std::unique_ptr<RtpReceiver> rtp_receiver_;
TelephoneEventHandler* telephone_event_handler_;
- rtc::scoped_ptr<RtpRtcp> _rtpRtcpModule;
- rtc::scoped_ptr<AudioCodingModule> audio_coding_;
- rtc::scoped_ptr<AudioSinkInterface> audio_sink_;
+ std::unique_ptr<RtpRtcp> _rtpRtcpModule;
+ std::unique_ptr<AudioCodingModule> audio_coding_;
+ std::unique_ptr<AudioSinkInterface> audio_sink_;
AudioLevel _outputAudioLevel;
bool _externalTransport;
AudioFrame _audioFrame;
@@ -535,7 +536,7 @@
rtc::CriticalSection ts_stats_lock_;
- rtc::scoped_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
+ std::unique_ptr<rtc::TimestampWrapAroundHandler> rtp_ts_wraparound_handler_;
// The rtp timestamp of the first played out audio frame.
int64_t capture_start_rtp_time_stamp_;
// The capture ntp time (in local timebase) of the first played out audio
@@ -552,7 +553,7 @@
rtc::CriticalSection* _callbackCritSectPtr; // owned by base
Transport* _transportPtr; // WebRtc socket or external transport
RMSLevel rms_level_;
- rtc::scoped_ptr<AudioProcessing> rx_audioproc_; // far end AudioProcessing
+ std::unique_ptr<AudioProcessing> rx_audioproc_; // far end AudioProcessing
VoERxVadCallback* _rxVadObserverPtr;
int32_t _oldVadDecision;
int32_t _sendFrameType; // Send data is voice, 1-voice, 0-otherwise
@@ -584,17 +585,17 @@
bool _rxNsIsEnabled;
bool restored_packet_in_use_;
// RtcpBandwidthObserver
- rtc::scoped_ptr<VoERtcpObserver> rtcp_observer_;
- rtc::scoped_ptr<NetworkPredictor> network_predictor_;
+ std::unique_ptr<VoERtcpObserver> rtcp_observer_;
+ std::unique_ptr<NetworkPredictor> network_predictor_;
// An associated send channel.
rtc::CriticalSection assoc_send_channel_lock_;
ChannelOwner associate_send_channel_ GUARDED_BY(assoc_send_channel_lock_);
bool pacing_enabled_;
PacketRouter* packet_router_ = nullptr;
- rtc::scoped_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
- rtc::scoped_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
- rtc::scoped_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
+ std::unique_ptr<TransportFeedbackProxy> feedback_observer_proxy_;
+ std::unique_ptr<TransportSequenceNumberProxy> seq_num_allocator_proxy_;
+ std::unique_ptr<RtpPacketSenderProxy> rtp_packet_sender_proxy_;
};
} // namespace voe
diff --git a/voice_engine/channel_manager.cc b/voice_engine/channel_manager.cc
index eac2e50..96f6d2b 100644
--- a/voice_engine/channel_manager.cc
+++ b/voice_engine/channel_manager.cc
@@ -49,7 +49,7 @@
: instance_id_(instance_id),
last_channel_id_(-1),
config_(config),
- event_log_(RtcEventLog::Create()) {}
+ event_log_(rtc::ScopedToUnique(RtcEventLog::Create())) {}
ChannelOwner ChannelManager::CreateChannel() {
return CreateChannelInternal(config_);
diff --git a/voice_engine/channel_manager.h b/voice_engine/channel_manager.h
index 7c86959..77dfc45 100644
--- a/voice_engine/channel_manager.h
+++ b/voice_engine/channel_manager.h
@@ -11,11 +11,11 @@
#ifndef WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_H
#define WEBRTC_VOICE_ENGINE_CHANNEL_MANAGER_H
+#include <memory>
#include <vector>
#include "webrtc/base/constructormagic.h"
#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/typedefs.h"
@@ -62,7 +62,7 @@
// deleted when no references to them are held.
struct ChannelRef {
ChannelRef(Channel* channel);
- const rtc::scoped_ptr<Channel> channel;
+ const std::unique_ptr<Channel> channel;
Atomic32 ref_count;
};
@@ -127,7 +127,7 @@
std::vector<ChannelOwner> channels_;
const Config& config_;
- rtc::scoped_ptr<RtcEventLog> event_log_;
+ std::unique_ptr<RtcEventLog> event_log_;
RTC_DISALLOW_COPY_AND_ASSIGN(ChannelManager);
};
diff --git a/voice_engine/channel_proxy.cc b/voice_engine/channel_proxy.cc
index 1e2281a..3beaf9b 100644
--- a/voice_engine/channel_proxy.cc
+++ b/voice_engine/channel_proxy.cc
@@ -155,7 +155,7 @@
channel()->SendTelephoneEventOutband(event, duration_ms, 10, false) == 0;
}
-void ChannelProxy::SetSink(rtc::scoped_ptr<AudioSinkInterface> sink) {
+void ChannelProxy::SetSink(std::unique_ptr<AudioSinkInterface> sink) {
RTC_DCHECK(thread_checker_.CalledOnValidThread());
channel()->SetSink(std::move(sink));
}
diff --git a/voice_engine/channel_proxy.h b/voice_engine/channel_proxy.h
index 9d6839c..3461cf3 100644
--- a/voice_engine/channel_proxy.h
+++ b/voice_engine/channel_proxy.h
@@ -15,6 +15,7 @@
#include "webrtc/voice_engine/channel_manager.h"
#include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
+#include <memory>
#include <string>
#include <vector>
@@ -69,7 +70,7 @@
virtual bool SetSendTelephoneEventPayloadType(int payload_type);
virtual bool SendTelephoneEventOutband(uint8_t event, uint32_t duration_ms);
- virtual void SetSink(rtc::scoped_ptr<AudioSinkInterface> sink);
+ virtual void SetSink(std::unique_ptr<AudioSinkInterface> sink);
private:
Channel* channel() const;
diff --git a/voice_engine/network_predictor.h b/voice_engine/network_predictor.h
index b35ccd8..bf08fe9 100644
--- a/voice_engine/network_predictor.h
+++ b/voice_engine/network_predictor.h
@@ -11,6 +11,8 @@
#ifndef WEBRTC_VOICE_ENGINE_NETWORK_PREDICTOR_H_
#define WEBRTC_VOICE_ENGINE_NETWORK_PREDICTOR_H_
+#include <memory>
+
#include "webrtc/base/exp_filter.h"
#include "webrtc/system_wrappers/include/clock.h"
@@ -38,7 +40,7 @@
int64_t last_loss_rate_update_time_ms_;
// An exponential filter is used to predict packet loss rate.
- rtc::scoped_ptr<rtc::ExpFilter> loss_rate_filter_;
+ std::unique_ptr<rtc::ExpFilter> loss_rate_filter_;
};
} // namespace voe
diff --git a/voice_engine/network_predictor_unittest.cc b/voice_engine/network_predictor_unittest.cc
index 28ff57f..1471f46 100644
--- a/voice_engine/network_predictor_unittest.cc
+++ b/voice_engine/network_predictor_unittest.cc
@@ -10,6 +10,8 @@
#include <math.h>
+#include <memory>
+
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/voice_engine/network_predictor.h"
#include "webrtc/system_wrappers/include/clock.h"
@@ -23,7 +25,7 @@
: clock_(0),
network_predictor_(new NetworkPredictor(&clock_)) {}
SimulatedClock clock_;
- rtc::scoped_ptr<NetworkPredictor> network_predictor_;
+ std::unique_ptr<NetworkPredictor> network_predictor_;
};
TEST_F(TestNetworkPredictor, TestPacketLossRateFilter) {
diff --git a/voice_engine/shared_data.cc b/voice_engine/shared_data.cc
index 1d50c3c..b21578c 100644
--- a/voice_engine/shared_data.cc
+++ b/voice_engine/shared_data.cc
@@ -27,7 +27,8 @@
_channelManager(_gInstanceCounter, config),
_engineStatistics(_gInstanceCounter),
_audioDevicePtr(NULL),
- _moduleProcessThreadPtr(ProcessThread::Create("VoiceProcessThread")) {
+ _moduleProcessThreadPtr(
+ rtc::ScopedToUnique(ProcessThread::Create("VoiceProcessThread"))) {
Trace::CreateTrace();
if (OutputMixer::Create(_outputMixerPtr, _gInstanceCounter) == 0)
{
diff --git a/voice_engine/shared_data.h b/voice_engine/shared_data.h
index 1d96103..a4a852a 100644
--- a/voice_engine/shared_data.h
+++ b/voice_engine/shared_data.h
@@ -11,8 +11,9 @@
#ifndef WEBRTC_VOICE_ENGINE_SHARED_DATA_H
#define WEBRTC_VOICE_ENGINE_SHARED_DATA_H
+#include <memory>
+
#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_device/include/audio_device.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
#include "webrtc/modules/utility/include/process_thread.h"
@@ -69,8 +70,8 @@
AudioDeviceModule* _audioDevicePtr;
OutputMixer* _outputMixerPtr;
TransmitMixer* _transmitMixerPtr;
- rtc::scoped_ptr<AudioProcessing> audioproc_;
- rtc::scoped_ptr<ProcessThread> _moduleProcessThreadPtr;
+ std::unique_ptr<AudioProcessing> audioproc_;
+ std::unique_ptr<ProcessThread> _moduleProcessThreadPtr;
AudioDeviceModule::AudioLayer _audioDeviceLayer;
diff --git a/voice_engine/test/auto_test/fakes/conference_transport.h b/voice_engine/test/auto_test/fakes/conference_transport.h
index cce6148..8fd7457 100644
--- a/voice_engine/test/auto_test/fakes/conference_transport.h
+++ b/voice_engine/test/auto_test/fakes/conference_transport.h
@@ -13,13 +13,13 @@
#include <deque>
#include <map>
+#include <memory>
#include <utility>
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/basictypes.h"
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
@@ -130,7 +130,7 @@
rtc::CriticalSection pq_crit_;
rtc::CriticalSection stream_crit_;
- const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
+ const std::unique_ptr<webrtc::EventWrapper> packet_event_;
rtc::PlatformThread thread_;
unsigned int rtt_ms_;
@@ -156,7 +156,7 @@
LoudestFilter loudest_filter_;
- const rtc::scoped_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
+ const std::unique_ptr<webrtc::RtpHeaderParser> rtp_header_parser_;
};
} // namespace voetest
diff --git a/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h b/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
index 46b7abf..6608669 100644
--- a/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
+++ b/voice_engine/test/auto_test/fixtures/after_initialization_fixture.h
@@ -12,10 +12,10 @@
#define SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
#include <deque>
+#include <memory>
#include "webrtc/base/criticalsection.h"
#include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/rtp_rtcp/source/byte_io.h"
#include "webrtc/system_wrappers/include/atomic32.h"
@@ -143,7 +143,7 @@
}
rtc::CriticalSection crit_;
- const rtc::scoped_ptr<webrtc::EventWrapper> packet_event_;
+ const std::unique_ptr<webrtc::EventWrapper> packet_event_;
rtc::PlatformThread thread_;
std::deque<Packet> packet_queue_ GUARDED_BY(crit_);
const int channel_;
@@ -163,7 +163,7 @@
virtual ~AfterInitializationFixture();
protected:
- rtc::scoped_ptr<TestErrorObserver> error_observer_;
+ std::unique_ptr<TestErrorObserver> error_observer_;
};
#endif // SRC_VOICE_ENGINE_MAIN_TEST_AUTO_TEST_STANDARD_TEST_BASE_AFTER_INIT_H_
diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
index 1dc15df..16f17b1 100644
--- a/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
+++ b/voice_engine/test/auto_test/standard/rtp_rtcp_extensions.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <memory>
+
#include "webrtc/modules/include/module_common_types.h"
#include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
#include "webrtc/system_wrappers/include/atomic32.h"
@@ -83,7 +85,7 @@
kPacketsExpected = 10,
kSleepIntervalMs = 10
};
- rtc::scoped_ptr<webrtc::RtpHeaderParser> parser_;
+ std::unique_ptr<webrtc::RtpHeaderParser> parser_;
webrtc::Atomic32 received_packets_;
webrtc::Atomic32 bad_packets_;
int audio_level_id_;
diff --git a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
index d653cc1..eeb7fa5 100644
--- a/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
+++ b/voice_engine/test/auto_test/standard/rtp_rtcp_test.cc
@@ -8,6 +8,8 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <memory>
+
#include "webrtc/base/criticalsection.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/system_wrappers/include/event_wrapper.h"
@@ -35,7 +37,7 @@
public:
rtc::CriticalSection crit_;
unsigned int incoming_ssrc_;
- rtc::scoped_ptr<voetest::EventWrapper> changed_ssrc_event_;
+ std::unique_ptr<voetest::EventWrapper> changed_ssrc_event_;
};
void TestRtpObserver::OnIncomingSSRCChanged(int channel,
diff --git a/voice_engine/test/auto_test/voe_cpu_test.cc b/voice_engine/test/auto_test/voe_cpu_test.cc
index 5666b3f..3bf51aa 100644
--- a/voice_engine/test/auto_test/voe_cpu_test.cc
+++ b/voice_engine/test/auto_test/voe_cpu_test.cc
@@ -17,7 +17,8 @@
#include <conio.h>
#endif
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
#include "webrtc/test/channel_transport/channel_transport.h"
#include "webrtc/voice_engine/test/auto_test/voe_test_defines.h"
@@ -64,7 +65,7 @@
CHECK(base->Init());
channel = base->CreateChannel();
- rtc::scoped_ptr<VoiceChannelTransport> voice_socket_transport(
+ std::unique_ptr<VoiceChannelTransport> voice_socket_transport(
new VoiceChannelTransport(voe_network, channel));
CHECK(voice_socket_transport->SetSendDestination("127.0.0.1", 5566));
diff --git a/voice_engine/test/auto_test/voe_output_test.cc b/voice_engine/test/auto_test/voe_output_test.cc
index 3bedbc3..d1bcf96 100644
--- a/voice_engine/test/auto_test/voe_output_test.cc
+++ b/voice_engine/test/auto_test/voe_output_test.cc
@@ -10,7 +10,6 @@
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/random.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/base/timeutils.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/channel_transport/channel_transport.h"
diff --git a/voice_engine/test/auto_test/voe_stress_test.cc b/voice_engine/test/auto_test/voe_stress_test.cc
index 259eff0..0b5660f 100644
--- a/voice_engine/test/auto_test/voe_stress_test.cc
+++ b/voice_engine/test/auto_test/voe_stress_test.cc
@@ -24,7 +24,6 @@
#include "webrtc/voice_engine/test/auto_test/voe_stress_test.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/system_wrappers/include/sleep.h"
#include "webrtc/test/channel_transport/channel_transport.h"
#include "webrtc/voice_engine/test/auto_test/voe_standard_test.h"
@@ -144,7 +143,7 @@
printf("Test will take approximately %d minutes. \n",
numberOfLoops * loopSleep / 1000 / 60 + 1);
- rtc::scoped_ptr<VoiceChannelTransport> voice_channel_transport(
+ std::unique_ptr<VoiceChannelTransport> voice_channel_transport(
new VoiceChannelTransport(voe_network, 0));
for (i = 0; i < numberOfLoops; ++i) {
diff --git a/voice_engine/test/auto_test/voe_stress_test.h b/voice_engine/test/auto_test/voe_stress_test.h
index 715e8ef..1c91306 100644
--- a/voice_engine/test/auto_test/voe_stress_test.h
+++ b/voice_engine/test/auto_test/voe_stress_test.h
@@ -11,8 +11,9 @@
#ifndef WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
#define WEBRTC_VOICE_ENGINE_VOE_STRESS_TEST_H
+#include <memory>
+
#include "webrtc/base/platform_thread.h"
-#include "webrtc/base/scoped_ptr.h"
namespace voetest {
@@ -37,8 +38,8 @@
VoETestManager& _mgr;
- // TODO(pbos): Remove scoped_ptr and use PlatformThread directly.
- rtc::scoped_ptr<rtc::PlatformThread> _ptrExtraApiThread;
+ // TODO(pbos): Remove unique_ptr and use PlatformThread directly.
+ std::unique_ptr<rtc::PlatformThread> _ptrExtraApiThread;
};
} // namespace voetest
diff --git a/voice_engine/test/cmd_test/voe_cmd_test.cc b/voice_engine/test/cmd_test/voe_cmd_test.cc
index ccfe3c2..bfc8b65 100644
--- a/voice_engine/test/cmd_test/voe_cmd_test.cc
+++ b/voice_engine/test/cmd_test/voe_cmd_test.cc
@@ -15,12 +15,12 @@
#include <unistd.h>
#endif
+#include <memory>
#include <vector>
#include "gflags/gflags.h"
#include "testing/gtest/include/gtest/gtest.h"
#include "webrtc/base/format_macros.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/call/rtc_event_log.h"
#include "webrtc/engine_configurations.h"
#include "webrtc/modules/audio_processing/include/audio_processing.h"
@@ -142,7 +142,7 @@
MyObserver my_observer;
- rtc::scoped_ptr<test::TraceToStderr> trace_to_stderr;
+ std::unique_ptr<test::TraceToStderr> trace_to_stderr;
if (!FLAGS_use_log_file) {
trace_to_stderr.reset(new test::TraceToStderr);
} else {
diff --git a/voice_engine/transmit_mixer.cc b/voice_engine/transmit_mixer.cc
index 1490350..d6a5213 100644
--- a/voice_engine/transmit_mixer.cc
+++ b/voice_engine/transmit_mixer.cc
@@ -10,6 +10,8 @@
#include "webrtc/voice_engine/transmit_mixer.h"
+#include <memory>
+
#include "webrtc/base/format_macros.h"
#include "webrtc/base/logging.h"
#include "webrtc/modules/utility/include/audio_frame_operations.h"
@@ -1180,7 +1182,7 @@
int32_t TransmitMixer::MixOrReplaceAudioWithFile(
int mixingFrequency)
{
- rtc::scoped_ptr<int16_t[]> fileBuffer(new int16_t[640]);
+ std::unique_ptr<int16_t[]> fileBuffer(new int16_t[640]);
size_t fileSamples(0);
{
diff --git a/voice_engine/transmit_mixer.h b/voice_engine/transmit_mixer.h
index e0246cf..b5c483a 100644
--- a/voice_engine/transmit_mixer.h
+++ b/voice_engine/transmit_mixer.h
@@ -12,7 +12,6 @@
#define WEBRTC_VOICE_ENGINE_TRANSMIT_MIXER_H
#include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/common_audio/resampler/include/push_resampler.h"
#include "webrtc/common_types.h"
#include "webrtc/modules/audio_processing/typing_detection.h"
diff --git a/voice_engine/voe_codec_unittest.cc b/voice_engine/voe_codec_unittest.cc
index f09e19e..73e576b 100644
--- a/voice_engine/voe_codec_unittest.cc
+++ b/voice_engine/voe_codec_unittest.cc
@@ -8,10 +8,11 @@
* be found in the AUTHORS file in the root of the source tree.
*/
+#include <memory>
+
#include "webrtc/voice_engine/include/voe_codec.h"
#include "testing/gtest/include/gtest/gtest.h"
-#include "webrtc/base/scoped_ptr.h"
#include "webrtc/modules/audio_device/include/fake_audio_device.h"
#include "webrtc/voice_engine/include/voe_base.h"
#include "webrtc/voice_engine/include/voe_hardware.h"
@@ -93,7 +94,7 @@
int channel_;
CodecInst primary_;
CodecInst valid_secondary_;
- rtc::scoped_ptr<FakeAudioDeviceModule> adm_;
+ std::unique_ptr<FakeAudioDeviceModule> adm_;
// A codec which is not valid to be registered as secondary codec.
CodecInst invalid_secondary_;
diff --git a/voice_engine/voice_engine_impl.cc b/voice_engine/voice_engine_impl.cc
index d0b7412..919d0e8 100644
--- a/voice_engine/voice_engine_impl.cc
+++ b/voice_engine/voice_engine_impl.cc
@@ -62,12 +62,12 @@
return new_ref;
}
-rtc::scoped_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
+std::unique_ptr<voe::ChannelProxy> VoiceEngineImpl::GetChannelProxy(
int channel_id) {
RTC_DCHECK(channel_id >= 0);
rtc::CritScope cs(crit_sec());
RTC_DCHECK(statistics().Initialized());
- return rtc::scoped_ptr<voe::ChannelProxy>(
+ return std::unique_ptr<voe::ChannelProxy>(
new voe::ChannelProxy(channel_manager().GetChannel(channel_id)));
}
diff --git a/voice_engine/voice_engine_impl.h b/voice_engine/voice_engine_impl.h
index f98f881..ed6efe3 100644
--- a/voice_engine/voice_engine_impl.h
+++ b/voice_engine/voice_engine_impl.h
@@ -11,7 +11,8 @@
#ifndef WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
#define WEBRTC_VOICE_ENGINE_VOICE_ENGINE_IMPL_H
-#include "webrtc/base/scoped_ptr.h"
+#include <memory>
+
#include "webrtc/engine_configurations.h"
#include "webrtc/system_wrappers/include/atomic32.h"
#include "webrtc/voice_engine/voe_base_impl.h"
@@ -134,14 +135,14 @@
// Backdoor to access a voe::Channel object without a channel ID. This is only
// to be used while refactoring the VoE API!
- virtual rtc::scoped_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id);
+ virtual std::unique_ptr<voe::ChannelProxy> GetChannelProxy(int channel_id);
// This is *protected* so that FakeVoiceEngine can inherit from the class and
// manipulate the reference count. See: fake_voice_engine.h.
protected:
Atomic32 _ref_count;
private:
- rtc::scoped_ptr<const Config> own_config_;
+ std::unique_ptr<const Config> own_config_;
};
} // namespace webrtc
