Refactor FakeAudioDevice to have separate methods for starting recording and playout.

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 128 matching lines @@
                                       float video_ntp_speed,
                                       float video_rtp_speed,
                                       float audio_rtp_speed) {
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
 
   metrics::Reset();
   VoiceEngine* voice_engine = VoiceEngine::Create();
   VoEBase* voe_base = VoEBase::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);
+  FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_rtp_speed,
+                                    48000);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
   int send_channel_id = voe_base->CreateChannel(config);
   int 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();
   Call::Config sender_config(&event_log_);
@@ skipping 93 matching lines @@
   }
   EXPECT_EQ(1u, video_receive_streams_.size());
   observer.set_receive_stream(video_receive_streams_[0]);
   DriftingClock drifting_clock(clock_, video_ntp_speed);
   CreateFrameGeneratorCapturerWithDrift(&drifting_clock, video_rtp_speed,
                                         kDefaultFramerate, kDefaultWidth,
                                         kDefaultHeight);
 
   Start();
 
-  fake_audio_device.Start();
+  fake_audio_device.StartRecordingPulsedNoise(256);
+  fake_audio_device.StartPlayout();
   audio_receive_stream->Start();
   EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
 
   EXPECT_TRUE(observer.Wait())
       << "Timed out while waiting for audio and video to be synchronized.";
 
   EXPECT_EQ(0, voe_base->StopSend(send_channel_id));
   EXPECT_EQ(0, voe_base->StopPlayout(recv_channel_id));
-  fake_audio_device.Stop();
+  fake_audio_device.StopRecording();
+  fake_audio_device.StopPlayout();
 
   Stop();
   video_send_transport.StopSending();
   audio_send_transport.StopSending();
   receive_transport.StopSending();
 
   DestroyStreams();
 
   sender_call_->DestroyAudioSendStream(audio_send_stream);
   receiver_call_->DestroyAudioReceiveStream(audio_receive_stream);
@@ skipping 446 matching lines @@
     uint32_t last_set_bitrate_kbps_;
     VideoSendStream* send_stream_;
     test::FrameGeneratorCapturer* frame_generator_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc