Making FakeNetworkPipe demux audio and video packets.

webrtc/call/call_perf_tests.cc

Index: webrtc/call/call_perf_tests.cc
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index a31cb23922e4aecd82a96a7f71a0b2c971de0238..32eea24c465ab7d38be83a32178ce46626a24ef7 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -38,6 +38,7 @@
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/video/transport_adapter.h"
+#include "webrtc/video/video_quality_test.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 
 using webrtc::test::DriftingClock;
@@ -167,27 +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;