Replace scoped_ptr with unique_ptr in webrtc/call/

BUG=webrtc:5520

Review URL: https://codereview.webrtc.org/1789903003

Cr-Commit-Position: refs/heads/master@{#11970}

webrtc/call/bitrate_allocator.h

@@ -11,12 +11,13 @@
 #ifndef WEBRTC_CALL_BITRATE_ALLOCATOR_H_
 #define WEBRTC_CALL_BITRATE_ALLOCATOR_H_
 
+#include <stdint.h>
+
 #include <list>
 #include <map>
 #include <utility>
 
 #include "webrtc/base/criticalsection.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 
 namespace webrtc {

webrtc/call/bitrate_allocator_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include <algorithm>
+#include <memory>
 #include <vector>
 
 #include "testing/gtest/include/gtest/gtest.h"
@@ -41,7 +42,7 @@ class BitrateAllocatorTest : public ::testing::Test {
   }
   ~BitrateAllocatorTest() {}
 
-  rtc::scoped_ptr<BitrateAllocator> allocator_;
+  std::unique_ptr<BitrateAllocator> allocator_;
 };
 
 TEST_F(BitrateAllocatorTest, UpdatingBitrateObserver) {
@@ -105,7 +106,7 @@ class BitrateAllocatorTestNoEnforceMin : public ::testing::Test {
   }
   ~BitrateAllocatorTestNoEnforceMin() {}
 
-  rtc::scoped_ptr<BitrateAllocator> allocator_;
+  std::unique_ptr<BitrateAllocator> allocator_;
 };
 
 // The following three tests verify that the EnforceMinBitrate() method works

webrtc/call/bitrate_estimator_tests.cc

@@ -9,6 +9,7 @@
  */
 #include <functional>
 #include <list>
+#include <memory>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
@@ -17,7 +18,6 @@
 #include "webrtc/base/checks.h"
 #include "webrtc/base/event.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
@@ -243,17 +243,17 @@ class BitrateEstimatorTest : public test::CallTest {
     VideoSendStream* send_stream_;
     AudioReceiveStream* audio_receive_stream_;
     VideoReceiveStream* video_receive_stream_;
-    rtc::scoped_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
+    std::unique_ptr<test::FrameGeneratorCapturer> frame_generator_capturer_;
     test::FakeEncoder fake_encoder_;
     test::FakeDecoder fake_decoder_;
   };
 
   testing::NiceMock<test::MockVoiceEngine> mock_voice_engine_;
   LogObserver receiver_log_;
-  rtc::scoped_ptr<test::DirectTransport> send_transport_;
-  rtc::scoped_ptr<test::DirectTransport> receive_transport_;
-  rtc::scoped_ptr<Call> sender_call_;
-  rtc::scoped_ptr<Call> receiver_call_;
+  std::unique_ptr<test::DirectTransport> send_transport_;
+  std::unique_ptr<test::DirectTransport> receive_transport_;
+  std::unique_ptr<Call> sender_call_;
+  std::unique_ptr<Call> receiver_call_;
   VideoReceiveStream::Config receive_config_;
   std::vector<Stream*> streams_;
 };

webrtc/call/call.cc

@@ -11,6 +11,7 @@
 #include <string.h>
 
 #include <map>
+#include <memory>
 #include <vector>
 
 #include "webrtc/audio/audio_receive_stream.h"
@@ -19,7 +20,6 @@
 #include "webrtc/audio/scoped_voe_interface.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/base/thread_checker.h"
 #include "webrtc/base/trace_event.h"
@@ -120,16 +120,16 @@ class Call : public webrtc::Call, public PacketReceiver,
   Clock* const clock_;
 
   const int num_cpu_cores_;
-  const rtc::scoped_ptr<ProcessThread> module_process_thread_;
-  const rtc::scoped_ptr<ProcessThread> pacer_thread_;
-  const rtc::scoped_ptr<CallStats> call_stats_;
-  const rtc::scoped_ptr<BitrateAllocator> bitrate_allocator_;
+  const std::unique_ptr<ProcessThread> module_process_thread_;
+  const std::unique_ptr<ProcessThread> pacer_thread_;
+  const std::unique_ptr<CallStats> call_stats_;
+  const std::unique_ptr<BitrateAllocator> bitrate_allocator_;
   Call::Config config_;
   rtc::ThreadChecker configuration_thread_checker_;
 
   bool network_enabled_;
 
-  rtc::scoped_ptr<RWLockWrapper> receive_crit_;
+  std::unique_ptr<RWLockWrapper> receive_crit_;
   // Audio and Video receive streams are owned by the client that creates them.
   std::map<uint32_t, AudioReceiveStream*> audio_receive_ssrcs_
       GUARDED_BY(receive_crit_);
@@ -140,7 +140,7 @@ class Call : public webrtc::Call, public PacketReceiver,
   std::map<std::string, AudioReceiveStream*> sync_stream_mapping_
       GUARDED_BY(receive_crit_);
 
-  rtc::scoped_ptr<RWLockWrapper> send_crit_;
+  std::unique_ptr<RWLockWrapper> send_crit_;
   // Audio and Video send streams are owned by the client that creates them.
   std::map<uint32_t, AudioSendStream*> audio_send_ssrcs_ GUARDED_BY(send_crit_);
   std::map<uint32_t, VideoSendStream*> video_send_ssrcs_ GUARDED_BY(send_crit_);
@@ -168,7 +168,7 @@ class Call : public webrtc::Call, public PacketReceiver,
   int64_t num_bitrate_updates_ GUARDED_BY(&bitrate_crit_);
 
   VieRemb remb_;
-  const rtc::scoped_ptr<CongestionController> congestion_controller_;
+  const std::unique_ptr<CongestionController> congestion_controller_;
 
   RTC_DISALLOW_COPY_AND_ASSIGN(Call);
 };
@@ -183,8 +183,9 @@ namespace internal {
 Call::Call(const Call::Config& config)
     : clock_(Clock::GetRealTimeClock()),
       num_cpu_cores_(CpuInfo::DetectNumberOfCores()),
-      module_process_thread_(ProcessThread::Create("ModuleProcessThread")),
-      pacer_thread_(ProcessThread::Create("PacerThread")),
+      module_process_thread_(
+          rtc::ScopedToUnique(ProcessThread::Create("ModuleProcessThread"))),
+      pacer_thread_(rtc::ScopedToUnique(ProcessThread::Create("PacerThread"))),
       call_stats_(new CallStats(clock_)),
       bitrate_allocator_(new BitrateAllocator()),
       config_(config),

webrtc/call/call_perf_tests.cc

@@ -8,13 +8,13 @@
  *  be found in the AUTHORS file in the root of the source tree.
  */
 #include <algorithm>
+#include <memory>
 #include <sstream>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
@@ -235,7 +235,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
    private:
     int channel_;
     VoENetwork* voe_network_;
-    rtc::scoped_ptr<RtpHeaderParser> parser_;
+    std::unique_ptr<RtpHeaderParser> parser_;
   };
 
   VoiceEngine* voice_engine = VoiceEngine::Create();

webrtc/call/call_unittest.cc

@@ -9,6 +9,7 @@
  */
 
 #include <list>
+#include <memory>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
@@ -31,7 +32,7 @@ struct CallHelper {
 
  private:
   testing::NiceMock<webrtc::test::MockVoiceEngine> voice_engine_;
-  rtc::scoped_ptr<webrtc::Call> call_;
+  std::unique_ptr<webrtc::Call> call_;
 };
 }  // namespace
 

webrtc/call/packet_injection_tests.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/test/call_test.h"
@@ -29,7 +31,7 @@ class PacketInjectionTest : public test::CallTest {
                              const uint8_t* packet,
                              size_t length);
 
-  rtc::scoped_ptr<RtpHeaderParser> rtp_header_parser_;
+  std::unique_ptr<RtpHeaderParser> rtp_header_parser_;
 };
 
 void PacketInjectionTest::InjectIncorrectPacket(CodecType codec_type,

webrtc/call/rampup_tests.h

@@ -16,7 +16,6 @@
 #include <vector>
 
 #include "webrtc/base/event.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call.h"
 #include "webrtc/test/call_test.h"
 

webrtc/call/rtc_event_log.cc

@@ -105,8 +105,8 @@ class RtcEventLogImpl final : public RtcEventLog {
       EXCLUSIVE_LOCKS_REQUIRED(crit_);
 
   rtc::CriticalSection crit_;
-  rtc::scoped_ptr<FileWrapper> file_ GUARDED_BY(crit_) =
-      rtc::scoped_ptr<FileWrapper>(FileWrapper::Create());
+  std::unique_ptr<FileWrapper> file_ GUARDED_BY(crit_) =
+      std::unique_ptr<FileWrapper>(FileWrapper::Create());
   rtc::PlatformFile platform_file_ GUARDED_BY(crit_) =
       rtc::kInvalidPlatformFileValue;
   rtclog::EventStream stream_ GUARDED_BY(crit_);
@@ -501,7 +501,7 @@ bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
                                    rtclog::EventStream* result) {
   char tmp_buffer[1024];
   int bytes_read = 0;
-  rtc::scoped_ptr<FileWrapper> dump_file(FileWrapper::Create());
+  std::unique_ptr<FileWrapper> dump_file(FileWrapper::Create());
   if (dump_file->OpenFile(file_name.c_str(), true) != 0) {
     return false;
   }
@@ -516,8 +516,8 @@ bool RtcEventLog::ParseRtcEventLog(const std::string& file_name,
 #endif  // ENABLE_RTC_EVENT_LOG
 
 // RtcEventLog member functions.
-rtc::scoped_ptr<RtcEventLog> RtcEventLog::Create() {
-  return rtc::scoped_ptr<RtcEventLog>(new RtcEventLogImpl());
+std::unique_ptr<RtcEventLog> RtcEventLog::Create() {
+  return std::unique_ptr<RtcEventLog>(new RtcEventLogImpl());
 }
 
 }  // namespace webrtc

webrtc/call/rtc_event_log.h

@@ -11,10 +11,10 @@
 #ifndef WEBRTC_CALL_RTC_EVENT_LOG_H_
 #define WEBRTC_CALL_RTC_EVENT_LOG_H_
 
+#include <memory>
 #include <string>
 
 #include "webrtc/base/platform_file.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/video_receive_stream.h"
 #include "webrtc/video_send_stream.h"
 
@@ -36,7 +36,7 @@ class RtcEventLog {
  public:
   virtual ~RtcEventLog() {}
 
-  static rtc::scoped_ptr<RtcEventLog> Create();
+  static std::unique_ptr<RtcEventLog> Create();
 
   // Sets the time that events are stored in the internal event buffer
   // before the user calls StartLogging.  The default is 10 000 000 us = 10 s

webrtc/call/rtc_event_log2rtp_dump.cc

@@ -9,12 +9,12 @@
  */
 
 #include <iostream>
+#include <memory>
 #include <sstream>
 #include <string>
 
 #include "gflags/gflags.h"
 #include "webrtc/base/checks.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/call/rtc_event_log.h"
 #include "webrtc/modules/rtp_rtcp/source/byte_io.h"
 #include "webrtc/test/rtp_file_writer.h"
@@ -100,7 +100,7 @@ int main(int argc, char* argv[]) {
     return -1;
   }
 
-  rtc::scoped_ptr<webrtc::test::RtpFileWriter> rtp_writer(
+  std::unique_ptr<webrtc::test::RtpFileWriter> rtp_writer(
       webrtc::test::RtpFileWriter::Create(
           webrtc::test::RtpFileWriter::FileFormat::kRtpDump, output_file));
 

webrtc/call/rtc_event_log_unittest.cc

@@ -10,6 +10,7 @@
 
 #ifdef ENABLE_RTC_EVENT_LOG
 
+#include <memory>
 #include <string>
 #include <utility>
 #include <vector>
@@ -18,7 +19,6 @@
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/checks.h"
 #include "webrtc/base/random.h"
-#include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread.h"
 #include "webrtc/call.h"
 #include "webrtc/call/rtc_event_log.h"
@@ -473,7 +473,7 @@ void LogSessionAndReadBack(size_t rtp_count,
   // When log_dumper goes out of scope, it causes the log file to be flushed
   // to disk.
   {
-    rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
+    std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
     log_dumper->LogVideoReceiveStreamConfig(receiver_config);
     log_dumper->LogVideoSendStreamConfig(sender_config);
     size_t rtcp_index = 1;
@@ -639,7 +639,7 @@ void DropOldEvents(uint32_t extensions_bitvector,
 
   // The log file will be flushed to disk when the log_dumper goes out of scope.
   {
-    rtc::scoped_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
+    std::unique_ptr<RtcEventLog> log_dumper(RtcEventLog::Create());
     // Reduce the time old events are stored to 50 ms.
     log_dumper->SetBufferDuration(50000);
     log_dumper->LogVideoReceiveStreamConfig(receiver_config);

webrtc/voice_engine/channel_manager.cc

@@ -49,7 +49,7 @@ ChannelManager::ChannelManager(uint32_t instance_id, const Config& config)
     : instance_id_(instance_id),
       last_channel_id_(-1),
       config_(config),
-      event_log_(rtc::ScopedToUnique(RtcEventLog::Create())) {}
+      event_log_(RtcEventLog::Create()) {}
 
 ChannelOwner ChannelManager::CreateChannel() {
   return CreateChannelInternal(config_);