Removes injection of RtpTransportControllerSend from Call::Create.
Bug: webrtc:10365
Change-Id: Ie319611828116f8ffbb582d5ab2099240b26699e
Reviewed-on: https://webrtc-review.googlesource.com/c/124784
Commit-Queue: Sebastian Jansson <srte@webrtc.org>
Reviewed-by: Ilya Nikolaevskiy <ilnik@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26926}diff --git a/call/call.cc b/call/call.cc
index 5df97e9..6655e22 100644
--- a/call/call.cc
+++ b/call/call.cc
@@ -419,12 +419,6 @@
config.network_controller_factory, config.bitrate_config));
}
-Call* Call::Create(
- const Call::Config& config,
- std::unique_ptr<RtpTransportControllerSendInterface> transport_send) {
- return new internal::Call(config, std::move(transport_send));
-}
-
// This method here to avoid subclasses has to implement this method.
// Call perf test will use Internal::Call::CreateVideoSendStream() to inject
// FecController.
diff --git a/call/call.h b/call/call.h
index ab834f3..eb5596d 100644
--- a/call/call.h
+++ b/call/call.h
@@ -50,11 +50,6 @@
static Call* Create(const Call::Config& config);
- // Allows mocking |transport_send| for testing.
- static Call* Create(
- const Call::Config& config,
- std::unique_ptr<RtpTransportControllerSendInterface> transport_send);
-
virtual AudioSendStream* CreateAudioSendStream(
const AudioSendStream::Config& config) = 0;
diff --git a/test/call_test.cc b/test/call_test.cc
index b8334b5..44df9cb 100644
--- a/test/call_test.cc
+++ b/test/call_test.cc
@@ -18,7 +18,6 @@
#include "api/video/builtin_video_bitrate_allocator_factory.h"
#include "api/video_codecs/video_encoder_config.h"
#include "call/fake_network_pipe.h"
-#include "call/rtp_transport_controller_send.h"
#include "call/simulated_network.h"
#include "modules/audio_mixer/audio_mixer_impl.h"
#include "rtc_base/checks.h"
@@ -33,7 +32,6 @@
: clock_(Clock::GetRealTimeClock()),
send_event_log_(RtcEventLog::CreateNull()),
recv_event_log_(RtcEventLog::CreateNull()),
- sender_call_transport_controller_(nullptr),
audio_send_config_(/*send_transport=*/nullptr,
/*media_transport=*/nullptr),
audio_send_stream_(nullptr),
@@ -113,10 +111,6 @@
send_config.audio_state->audio_transport());
}
CreateSenderCall(send_config);
- if (sender_call_transport_controller_ != nullptr) {
- test->OnRtpTransportControllerSendCreated(
- sender_call_transport_controller_);
- }
if (test->ShouldCreateReceivers()) {
Call::Config recv_config(recv_event_log_.get());
test->ModifyReceiverBitrateConfig(&recv_config.bitrate_config);
@@ -221,20 +215,7 @@
}
void CallTest::CreateSenderCall(const Call::Config& config) {
- NetworkControllerFactoryInterface* injected_factory =
- config.network_controller_factory;
- if (injected_factory) {
- RTC_LOG(LS_INFO) << "Using injected network controller factory";
- } else {
- RTC_LOG(LS_INFO) << "Using default network controller factory";
- }
-
- std::unique_ptr<RtpTransportControllerSend> controller_send =
- absl::make_unique<RtpTransportControllerSend>(
- Clock::GetRealTimeClock(), config.event_log, injected_factory,
- config.bitrate_config);
- sender_call_transport_controller_ = controller_send.get();
- sender_call_.reset(Call::Create(config, std::move(controller_send)));
+ sender_call_.reset(Call::Create(config));
}
void CallTest::CreateReceiverCall(const Call::Config& config) {
@@ -761,9 +742,6 @@
void BaseTest::ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config) {
}
-void BaseTest::OnRtpTransportControllerSendCreated(
- RtpTransportControllerSend* controller) {}
-
void BaseTest::OnCallsCreated(Call* sender_call, Call* receiver_call) {}
test::PacketTransport* BaseTest::CreateSendTransport(
diff --git a/test/call_test.h b/test/call_test.h
index dbe8e07..052d4c6 100644
--- a/test/call_test.h
+++ b/test/call_test.h
@@ -20,7 +20,6 @@
#include "api/test/video/function_video_encoder_factory.h"
#include "api/video/video_bitrate_allocator_factory.h"
#include "call/call.h"
-#include "call/rtp_transport_controller_send.h"
#include "logging/rtc_event_log/rtc_event_log.h"
#include "modules/audio_device/include/test_audio_device.h"
#include "test/encoder_settings.h"
@@ -179,7 +178,6 @@
std::unique_ptr<webrtc::RtcEventLog> send_event_log_;
std::unique_ptr<webrtc::RtcEventLog> recv_event_log_;
std::unique_ptr<Call> sender_call_;
- RtpTransportControllerSend* sender_call_transport_controller_;
std::unique_ptr<PacketTransport> send_transport_;
std::vector<VideoSendStream::Config> video_send_configs_;
std::vector<VideoEncoderConfig> video_encoder_configs_;
@@ -254,8 +252,6 @@
virtual void ModifySenderBitrateConfig(BitrateConstraints* bitrate_config);
virtual void ModifyReceiverBitrateConfig(BitrateConstraints* bitrate_config);
- virtual void OnRtpTransportControllerSendCreated(
- RtpTransportControllerSend* controller);
virtual void OnCallsCreated(Call* sender_call, Call* receiver_call);
virtual test::PacketTransport* CreateSendTransport(
diff --git a/video/end_to_end_tests/probing_tests.cc b/video/end_to_end_tests/probing_tests.cc
index dba3c3e..df2caf9 100644
--- a/video/end_to_end_tests/probing_tests.cc
+++ b/video/end_to_end_tests/probing_tests.cc
@@ -210,11 +210,6 @@
send_stream_ = send_stream;
}
- void OnRtpTransportControllerSendCreated(
- RtpTransportControllerSend* transport_controller) override {
- transport_controller_ = transport_controller;
- }
-
test::PacketTransport* CreateSendTransport(
test::SingleThreadedTaskQueueForTesting* task_queue,
Call* sender_call) override {
@@ -250,8 +245,10 @@
// In order to speed up the test we can interrupt exponential
// probing by toggling the network availability. The alternative
// is to wait for it to time out (1000 ms).
- transport_controller_->OnNetworkAvailability(false);
- transport_controller_->OnNetworkAvailability(true);
+ sender_call_->GetTransportControllerSend()->OnNetworkAvailability(
+ false);
+ sender_call_->GetTransportControllerSend()->OnNetworkAvailability(
+ true);
++state_;
}
@@ -288,7 +285,6 @@
SimulatedNetwork* send_simulated_network_;
VideoSendStream* send_stream_;
VideoEncoderConfig* encoder_config_;
- RtpTransportControllerSend* transport_controller_;
};
bool success = false;
