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

webrtc/call/call_perf_tests.cc

Index: webrtc/call/call_perf_tests.cc
diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 782b45c9f273ffd6fe180fa59a2b49d2d5d914ee..6a21c62426726962502452a029f037ca1d77a1e8 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -146,11 +146,7 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   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(audio_rtp_speed, 48000, 256);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr, decoder_factory_));
   VoEBase::ChannelConfig config;
   config.enable_voice_pacing = true;
@@ -264,16 +260,15 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
 
   Start();
 
-  fake_audio_device.Start();
+  audio_send_stream->Start();
   audio_receive_stream->Start();
   EXPECT_EQ(0, voe_base->StartSend(send_channel_id));

[the sun, 2017/01/31 16:27:07]

This line shouldn't be needed.

[perkj_webrtc, 2017/01/31 19:11:07]

Done.

 
   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();
+  audio_send_stream->Stop();
+  audio_receive_stream->Stop();
 
   Stop();
   video_send_transport.StopSending();