Delete helper class MediaTypePacketReceiver.

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 150 matching lines @@
   send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
   Call::Config sender_config(event_log_.get());
   sender_config.audio_state = AudioState::Create(send_audio_state_config);
   Call::Config receiver_config(event_log_.get());
   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;
 
   std::map<uint8_t, MediaType> audio_pt_map;
   std::map<uint8_t, MediaType> video_pt_map;
   std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),
                std::inserter(audio_pt_map, audio_pt_map.end()),
                [](const std::pair<const uint8_t, MediaType>& pair) {
                  return pair.second == MediaType::AUDIO;
                });
   std::copy_if(std::begin(payload_type_map_), std::end(payload_type_map_),
                std::inserter(video_pt_map, video_pt_map.end()),
                [](const std::pair<const uint8_t, MediaType>& pair) {
                  return pair.second == MediaType::VIDEO;
                });
 
   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,
       payload_type_map_, FakeNetworkPipe::Config());
   receive_transport.SetReceiver(sender_call_->Receiver());
 
   test::FakeDecoder fake_decoder;
 
   CreateSendConfig(1, 0, 0, &video_send_transport);
   CreateMatchingReceiveConfigs(&receive_transport);
@@ skipping 545 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc