Moved creation of AudioDecoderFactory to inside PeerConnectionFactory.

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 140 matching lines @@
 
   metrics::Reset();
   VoiceEngine* voice_engine = VoiceEngine::Create();
   VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
   VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
   const std::string audio_filename =
       test::ResourcePath("voice_engine/audio_long16", "pcm");
   ASSERT_STRNE("", audio_filename.c_str());
   FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
                                     audio_rtp_speed);
-  EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
+  EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
   Config voe_config;
   voe_config.Set<VoicePacing>(new VoicePacing(true));
   int send_channel_id = voe_base->CreateChannel(voe_config);
   int recv_channel_id = voe_base->CreateChannel();
 
   AudioState::Config send_audio_state_config;
   send_audio_state_config.voice_engine = voice_engine;
   Call::Config sender_config;
   sender_config.audio_state = AudioState::Create(send_audio_state_config);
   Call::Config receiver_config;
@@ skipping 69 matching lines @@
   }
   video_receive_configs_[0].rtp.nack.rtp_history_ms = 1000;
   video_receive_configs_[0].renderer = &observer;
   video_receive_configs_[0].sync_group = kSyncGroup;
 
   AudioReceiveStream::Config audio_recv_config;
   audio_recv_config.rtp.remote_ssrc = kAudioSendSsrc;
   audio_recv_config.rtp.local_ssrc = kAudioRecvSsrc;
   audio_recv_config.voe_channel_id = recv_channel_id;
   audio_recv_config.sync_group = kSyncGroup;
+  audio_recv_config.decoder_factory = decoder_factory_;
 
   AudioReceiveStream* audio_receive_stream;
 
   if (create_first == CreateOrder::kAudioFirst) {
     audio_receive_stream =
         receiver_call_->CreateAudioReceiveStream(audio_recv_config);
     CreateVideoStreams();
   } else {
     CreateVideoStreams();
     audio_receive_stream =
@@ skipping 433 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc