diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc index 1e6f28d340..3b1de73e65 100644 --- a/webrtc/call/call_perf_tests.cc +++ b/webrtc/call/call_perf_tests.cc @@ -168,27 +168,6 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, VideoRtcpAndSyncObserver observer(Clock::GetRealTimeClock()); - // Helper class to ensure we deliver correct media_type to the receiving call. - class MediaTypePacketReceiver : public PacketReceiver { - public: - MediaTypePacketReceiver(PacketReceiver* packet_receiver, - MediaType media_type) - : packet_receiver_(packet_receiver), media_type_(media_type) {} - - DeliveryStatus DeliverPacket(MediaType media_type, - const uint8_t* packet, - size_t length, - const PacketTime& packet_time) override { - return packet_receiver_->DeliverPacket(media_type_, packet, length, - packet_time); - } - private: - PacketReceiver* packet_receiver_; - const MediaType media_type_; - - RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(MediaTypePacketReceiver); - }; - FakeNetworkPipe::Config audio_net_config; audio_net_config.queue_delay_ms = 500; audio_net_config.loss_percent = 5; @@ -209,16 +188,12 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec, test::PacketTransport audio_send_transport(sender_call_.get(), &observer, test::PacketTransport::kSender, audio_pt_map, audio_net_config); - MediaTypePacketReceiver audio_receiver(receiver_call_->Receiver(), - MediaType::AUDIO); - audio_send_transport.SetReceiver(&audio_receiver); + audio_send_transport.SetReceiver(receiver_call_->Receiver()); test::PacketTransport video_send_transport( sender_call_.get(), &observer, test::PacketTransport::kSender, video_pt_map, FakeNetworkPipe::Config()); - MediaTypePacketReceiver video_receiver(receiver_call_->Receiver(), - MediaType::VIDEO); - video_send_transport.SetReceiver(&video_receiver); + video_send_transport.SetReceiver(receiver_call_->Receiver()); test::PacketTransport receive_transport( receiver_call_.get(), &observer, test::PacketTransport::kReceiver,