Making FakeNetworkPipe demux audio and video packets.

webrtc/call/call_perf_tests.cc

 /*
  *  Copyright (c) 2013 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 20 matching lines @@
 #include "webrtc/test/fake_decoder.h"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/field_trial.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #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;
 using webrtc::test::FakeAudioDevice;
 
 namespace webrtc {
 
 class CallPerfTest : public test::CallTest {
  protected:
   enum class FecMode {
@@ skipping 109 matching lines @@
   send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
   Call::Config sender_config(&event_log_);
   sender_config.audio_state = AudioState::Create(send_audio_state_config);
   Call::Config receiver_config(&event_log_);
   receiver_config.audio_state = sender_config.audio_state;
   CreateCalls(sender_config, receiver_config);
 
 
   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;
   test::PacketTransport audio_send_transport(sender_call_.get(), &observer,
                                              test::PacketTransport::kSender,
                                              MediaType::AUDIO,
                                              audio_net_config);
   MediaTypePacketReceiver audio_receiver(receiver_call_->Receiver(),
                                          MediaType::AUDIO);
   audio_send_transport.SetReceiver(&audio_receiver);
@@ skipping 562 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc