| 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 bec696bd4ec472d2214998a9b65f605c961e2a94..1308657d0e465514d953b92245ce1091333c373a 100644
|
| --- a/webrtc/audio/audio_send_stream_unittest.cc
|
| +++ b/webrtc/audio/audio_send_stream_unittest.cc
|
| @@ -22,7 +22,7 @@
|
| #include "webrtc/modules/audio_processing/include/mock_audio_processing.h"
|
| #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/pacing/mock/mock_paced_sender.h"
|
| #include "webrtc/modules/rtp_rtcp/mocks/mock_rtcp_rtt_stats.h"
|
| #include "webrtc/modules/rtp_rtcp/mocks/mock_rtp_rtcp.h"
|
| #include "webrtc/rtc_base/ptr_util.h"
|
| @@ -139,8 +139,8 @@ struct ConfigHelper {
|
| &simulated_clock_,
|
| nullptr /* observer */,
|
| &event_log_,
|
| - &packet_router_)),
|
| - fake_transport_(&packet_router_, send_side_cc_.get()),
|
| + &pacer_)),
|
| + fake_transport_(&packet_router_, &pacer_, send_side_cc_.get()),
|
| bitrate_allocator_(&limit_observer_),
|
| worker_queue_("ConfigHelper_worker_queue"),
|
| audio_encoder_(nullptr) {
|
| @@ -335,6 +335,7 @@ struct ConfigHelper {
|
| AudioProcessing::AudioProcessingStatistics audio_processing_stats_;
|
| SimulatedClock simulated_clock_;
|
| PacketRouter packet_router_;
|
| + testing::NiceMock<MockPacedSender> pacer_;
|
| std::unique_ptr<SendSideCongestionController> send_side_cc_;
|
| FakeRtpTransportControllerSend fake_transport_;
|
| MockRtcEventLog event_log_;
|
|
|