Index: webrtc/audio/audio_send_stream_unittest.cc |
diff --git a/webrtc/audio/audio_send_stream_unittest.cc b/webrtc/audio/audio_send_stream_unittest.cc |
index 99e7bd5254a9672a3b5ae7fa4e4253cc63566931..c66efea674abc0d3d3b3faad78240dc6d22c0526 100644 |
--- a/webrtc/audio/audio_send_stream_unittest.cc |
+++ b/webrtc/audio/audio_send_stream_unittest.cc |
@@ -15,7 +15,9 @@ |
#include "webrtc/audio/audio_send_stream.h" |
#include "webrtc/audio/audio_state.h" |
#include "webrtc/audio/conversion.h" |
+#include "webrtc/base/ptr_util.h" |
#include "webrtc/base/task_queue.h" |
+#include "webrtc/call/fake_rtp_transport_controller_send.h" |
#include "webrtc/call/rtp_transport_controller_send_interface.h" |
#include "webrtc/logging/rtc_event_log/mock/mock_rtc_event_log.h" |
#include "webrtc/modules/audio_mixer/audio_mixer_impl.h" |
@@ -23,8 +25,8 @@ |
#include "webrtc/modules/congestion_controller/include/mock/mock_congestion_observer.h" |
#include "webrtc/modules/congestion_controller/include/send_side_congestion_controller.h" |
#include "webrtc/modules/pacing/paced_sender.h" |
-#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
#include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h" |
+#include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h" |
#include "webrtc/test/gtest.h" |
#include "webrtc/test/mock_audio_encoder.h" |
#include "webrtc/test/mock_audio_encoder_factory.h" |
@@ -124,36 +126,15 @@ rtc::scoped_refptr<MockAudioEncoderFactory> SetupEncoderFactoryMock() { |
} |
struct ConfigHelper { |
- class FakeRtpTransportController |
- : public RtpTransportControllerSendInterface { |
- public: |
- explicit FakeRtpTransportController(RtcEventLog* event_log) |
- : simulated_clock_(123456), |
- send_side_cc_(&simulated_clock_, |
- &bitrate_observer_, |
- event_log, |
- &packet_router_) {} |
- PacketRouter* packet_router() override { return &packet_router_; } |
- |
- SendSideCongestionController* send_side_cc() override { |
- return &send_side_cc_; |
- } |
- TransportFeedbackObserver* transport_feedback_observer() override { |
- return &send_side_cc_; |
- } |
- |
- RtpPacketSender* packet_sender() override { return send_side_cc_.pacer(); } |
- |
- private: |
- SimulatedClock simulated_clock_; |
- testing::NiceMock<MockCongestionObserver> bitrate_observer_; |
- PacketRouter packet_router_; |
- SendSideCongestionController send_side_cc_; |
- }; |
- |
ConfigHelper(bool audio_bwe_enabled, bool expect_set_encoder_call) |
: stream_config_(nullptr), |
- fake_transport_(&event_log_), |
+ simulated_clock_(123456), |
+ send_side_cc_(rtc::MakeUnique<SendSideCongestionController>( |
+ &simulated_clock_, |
+ nullptr /* observer */, |
+ &event_log_, |
+ &packet_router_)), |
+ fake_transport_(send_side_cc_.get()), |
bitrate_allocator_(&limit_observer_), |
worker_queue_("ConfigHelper_worker_queue") { |
using testing::Invoke; |
@@ -322,11 +303,13 @@ struct ConfigHelper { |
rtc::scoped_refptr<AudioState> audio_state_; |
AudioSendStream::Config stream_config_; |
testing::StrictMock<MockVoEChannelProxy>* channel_proxy_ = nullptr; |
- testing::NiceMock<MockCongestionObserver> bitrate_observer_; |
MockAudioProcessing audio_processing_; |
MockTransmitMixer transmit_mixer_; |
AudioProcessing::AudioProcessingStatistics audio_processing_stats_; |
- FakeRtpTransportController fake_transport_; |
+ SimulatedClock simulated_clock_; |
+ PacketRouter packet_router_; |
+ std::unique_ptr<SendSideCongestionController> send_side_cc_; |
+ FakeRtpTransportControllerSend fake_transport_; |
MockRtcEventLog event_log_; |
MockRtpRtcp rtp_rtcp_; |
MockRtcpRttStats rtcp_rtt_stats_; |