| Index: webrtc/call/call.cc
|
| diff --git a/webrtc/call/call.cc b/webrtc/call/call.cc
|
| index 2dfb13a2e63b820cc04aa3223be17d29e066a7d2..3fd7a931d0cecc88cff0b30a47557db5a21ed586 100644
|
| --- a/webrtc/call/call.cc
|
| +++ b/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),
|
|
|