| 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 91201d67f66325be4bc0079785b93946d9add5a4..e70f21094f00ef324c640981130652fd92f78b3d 100644
|
| --- a/webrtc/audio/audio_send_stream_unittest.cc
|
| +++ b/webrtc/audio/audio_send_stream_unittest.cc
|
| @@ -14,7 +14,9 @@
|
| #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"
|
| @@ -70,36 +72,15 @@ class MockTransmitMixer : public voe::TransmitMixer {
|
| };
|
|
|
| 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_;
|
| - };
|
| -
|
| explicit ConfigHelper(bool audio_bwe_enabled)
|
| : 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;
|
| @@ -259,10 +240,12 @@ 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_;
|
| + SimulatedClock simulated_clock_;
|
| + PacketRouter packet_router_;
|
| + std::unique_ptr<SendSideCongestionController> send_side_cc_;
|
| FakeRtpTransportController fake_transport_;
|
| MockRtcEventLog event_log_;
|
| MockRtcpRttStats rtcp_rtt_stats_;
|
|
|