Replace `new rtc::RefCountedObject` with `rtc::make_ref_counted` in a few files
Bug: webrtc:12701
Change-Id: Ie50225374f811424faf20caf4cf454b2fd1c4dc9
Reviewed-on: https://webrtc-review.googlesource.com/c/src/+/215930
Reviewed-by: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Harald Alvestrand <hta@webrtc.org>
Commit-Queue: Tommi <tommi@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#33818}
diff --git a/call/adaptation/broadcast_resource_listener.cc b/call/adaptation/broadcast_resource_listener.cc
index 59bd1e0..876d4c0 100644
--- a/call/adaptation/broadcast_resource_listener.cc
+++ b/call/adaptation/broadcast_resource_listener.cc
@@ -83,8 +83,8 @@
MutexLock lock(&lock_);
RTC_DCHECK(is_listening_);
rtc::scoped_refptr<AdapterResource> adapter =
- new rtc::RefCountedObject<AdapterResource>(source_resource_->Name() +
- "Adapter");
+ rtc::make_ref_counted<AdapterResource>(source_resource_->Name() +
+ "Adapter");
adapters_.push_back(adapter);
return adapter;
}
diff --git a/call/adaptation/resource_adaptation_processor.cc b/call/adaptation/resource_adaptation_processor.cc
index 4925b64..741575a 100644
--- a/call/adaptation/resource_adaptation_processor.cc
+++ b/call/adaptation/resource_adaptation_processor.cc
@@ -72,7 +72,7 @@
VideoStreamAdapter* stream_adapter)
: task_queue_(nullptr),
resource_listener_delegate_(
- new rtc::RefCountedObject<ResourceListenerDelegate>(this)),
+ rtc::make_ref_counted<ResourceListenerDelegate>(this)),
resources_(),
stream_adapter_(stream_adapter),
last_reported_source_restrictions_(),
diff --git a/call/adaptation/test/fake_resource.cc b/call/adaptation/test/fake_resource.cc
index fa69e88..d125468 100644
--- a/call/adaptation/test/fake_resource.cc
+++ b/call/adaptation/test/fake_resource.cc
@@ -19,7 +19,7 @@
// static
rtc::scoped_refptr<FakeResource> FakeResource::Create(std::string name) {
- return new rtc::RefCountedObject<FakeResource>(name);
+ return rtc::make_ref_counted<FakeResource>(name);
}
FakeResource::FakeResource(std::string name)
diff --git a/call/call_perf_tests.cc b/call/call_perf_tests.cc
index 4cb9766..47d6e90 100644
--- a/call/call_perf_tests.cc
+++ b/call/call_perf_tests.cc
@@ -834,7 +834,7 @@
bitrate_allocator_factory_.get();
encoder_config->max_bitrate_bps = 2 * kReconfigureThresholdKbps * 1000;
encoder_config->video_stream_factory =
- new rtc::RefCountedObject<VideoStreamFactory>();
+ rtc::make_ref_counted<VideoStreamFactory>();
encoder_config_ = encoder_config->Copy();
}
diff --git a/call/call_unittest.cc b/call/call_unittest.cc
index d836362..b06af1e 100644
--- a/call/call_unittest.cc
+++ b/call/call_unittest.cc
@@ -50,14 +50,14 @@
task_queue_factory_ = webrtc::CreateDefaultTaskQueueFactory();
webrtc::AudioState::Config audio_state_config;
audio_state_config.audio_mixer =
- new rtc::RefCountedObject<webrtc::test::MockAudioMixer>();
+ rtc::make_ref_counted<webrtc::test::MockAudioMixer>();
audio_state_config.audio_processing =
use_null_audio_processing
? nullptr
- : new rtc::RefCountedObject<
+ : rtc::make_ref_counted<
NiceMock<webrtc::test::MockAudioProcessing>>();
audio_state_config.audio_device_module =
- new rtc::RefCountedObject<webrtc::test::MockAudioDeviceModule>();
+ rtc::make_ref_counted<webrtc::test::MockAudioDeviceModule>();
webrtc::Call::Config config(&event_log_);
config.audio_state = webrtc::AudioState::Create(audio_state_config);
config.task_queue_factory = task_queue_factory_.get();
@@ -118,7 +118,7 @@
config.rtp.remote_ssrc = 42;
config.rtcp_send_transport = &rtcp_send_transport;
config.decoder_factory =
- new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
+ rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* stream = call->CreateAudioReceiveStream(config);
EXPECT_NE(stream, nullptr);
call->DestroyAudioReceiveStream(stream);
@@ -157,7 +157,7 @@
MockTransport rtcp_send_transport;
config.rtcp_send_transport = &rtcp_send_transport;
config.decoder_factory =
- new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
+ rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>();
std::list<AudioReceiveStream*> streams;
for (int i = 0; i < 2; ++i) {
for (uint32_t ssrc = 0; ssrc < 1234567; ssrc += 34567) {
@@ -187,7 +187,7 @@
recv_config.rtp.local_ssrc = 777;
recv_config.rtcp_send_transport = &rtcp_send_transport;
recv_config.decoder_factory =
- new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
+ rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* recv_stream =
call->CreateAudioReceiveStream(recv_config);
EXPECT_NE(recv_stream, nullptr);
@@ -226,7 +226,7 @@
recv_config.rtp.local_ssrc = 777;
recv_config.rtcp_send_transport = &rtcp_send_transport;
recv_config.decoder_factory =
- new rtc::RefCountedObject<webrtc::MockAudioDecoderFactory>();
+ rtc::make_ref_counted<webrtc::MockAudioDecoderFactory>();
AudioReceiveStream* recv_stream =
call->CreateAudioReceiveStream(recv_config);
EXPECT_NE(recv_stream, nullptr);
diff --git a/call/rampup_tests.cc b/call/rampup_tests.cc
index 379f9dc..e2ea55b 100644
--- a/call/rampup_tests.cc
+++ b/call/rampup_tests.cc
@@ -160,7 +160,7 @@
encoder_config->number_of_streams = num_video_streams_;
encoder_config->max_bitrate_bps = 2000000;
encoder_config->video_stream_factory =
- new rtc::RefCountedObject<RampUpTester::VideoStreamFactory>();
+ rtc::make_ref_counted<RampUpTester::VideoStreamFactory>();
if (num_video_streams_ == 1) {
// For single stream rampup until 1mbps
expected_bitrate_bps_ = kSingleStreamTargetBps;
diff --git a/call/rtp_video_sender_unittest.cc b/call/rtp_video_sender_unittest.cc
index e8689e7..fd26f1c 100644
--- a/call/rtp_video_sender_unittest.cc
+++ b/call/rtp_video_sender_unittest.cc
@@ -891,7 +891,7 @@
TEST(RtpVideoSenderTest, SimulcastSenderRegistersFrameTransformers) {
rtc::scoped_refptr<MockFrameTransformer> transformer =
- new rtc::RefCountedObject<MockFrameTransformer>();
+ rtc::make_ref_counted<MockFrameTransformer>();
EXPECT_CALL(*transformer, RegisterTransformedFrameSinkCallback(_, kSsrc1));
EXPECT_CALL(*transformer, RegisterTransformedFrameSinkCallback(_, kSsrc2));
