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Unified Diff: webrtc/call/call.cc

Issue 1789903003: Replace scoped_ptr with unique_ptr in webrtc/call/ (Closed) Base URL: https://chromium.googlesource.com/external/webrtc.git@master
Patch Set: Created 4 years, 9 months ago
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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),

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