Removed an unused variable from CallPerfTest::TestAudioVideoSync()

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 11 matching lines @@
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/rtc_base/checks.h"
 #include "webrtc/rtc_base/constructormagic.h"
 #include "webrtc/rtc_base/thread_annotations.h"
 #include "webrtc/system_wrappers/include/metrics_default.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/drifting_clock.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
-#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/voice_engine/include/voe_base.h"
@@ skipping 155 matching lines @@
   test::PacketTransport video_send_transport(
       sender_call_.get(), &observer, test::PacketTransport::kSender,
       video_pt_map, FakeNetworkPipe::Config());
   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);
 
   AudioSendStream::Config audio_send_config(&audio_send_transport);
   audio_send_config.voe_channel_id = send_channel_id;
   audio_send_config.rtp.ssrc = kAudioSendSsrc;
   audio_send_config.send_codec_spec =
       rtc::Optional<AudioSendStream::Config::SendCodecSpec>(
           {kAudioSendPayloadType, {"ISAC", 16000, 1}});
   audio_send_config.encoder_factory = CreateBuiltinAudioEncoderFactory();
@@ skipping 537 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc