Reland of Add full stack tests for MediaCodec.

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 159 matching lines @@
 
   task_queue_.SendTask([&]() {
     metrics::Reset();
     audio_processing = AudioProcessing::Create();
     voice_engine = VoiceEngine::Create();
     voe_base = VoEBase::GetInterface(voice_engine);
     fake_audio_device = rtc::MakeUnique<FakeAudioDevice>(
         FakeAudioDevice::CreatePulsedNoiseCapturer(256, 48000),
         FakeAudioDevice::CreateDiscardRenderer(48000), audio_rtp_speed);
     EXPECT_EQ(0, voe_base->Init(fake_audio_device.get(), audio_processing.get(),
-                                decoder_factory_));
+                                audio_decoder_factory_));
     VoEBase::ChannelConfig config;
     config.enable_voice_pacing = true;
     send_channel_id = voe_base->CreateChannel(config);
     recv_channel_id = voe_base->CreateChannel();
 
     AudioState::Config send_audio_state_config;
     send_audio_state_config.voice_engine = voice_engine;
     send_audio_state_config.audio_mixer = AudioMixerImpl::Create();
     send_audio_state_config.audio_processing = audio_processing;
     Call::Config sender_config(event_log_.get());
@@ skipping 53 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_;
+    audio_recv_config.decoder_factory = audio_decoder_factory_;
     audio_recv_config.decoder_map = {
         {kAudioSendPayloadType, {"ISAC", 16000, 1}}};
 
     if (create_first == CreateOrder::kAudioFirst) {
       audio_receive_stream =
           receiver_call_->CreateAudioReceiveStream(audio_recv_config);
       CreateVideoStreams();
     } else {
       CreateVideoStreams();
       audio_receive_stream =
@@ skipping 514 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc