Added A/V sync tests with drifting clocks.

webrtc/test/fake_audio_device.cc

Index: webrtc/test/fake_audio_device.cc
diff --git a/webrtc/test/fake_audio_device.cc b/webrtc/test/fake_audio_device.cc
index 31cebda6521a01455fbe17f5b789423d296aa82e..0381981c207a191b74a669ef1881133f80208443 100644
--- a/webrtc/test/fake_audio_device.cc
+++ b/webrtc/test/fake_audio_device.cc
@@ -22,13 +22,17 @@
 namespace webrtc {
 namespace test {
 
-FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
+FakeAudioDevice::FakeAudioDevice(Clock* clock,
+                                 const std::string& filename,
+                                 float speed)
     : audio_callback_(NULL),
       capturing_(false),
       captured_audio_(),
       playout_buffer_(),
+      speed_(speed),
       last_playout_ms_(-1),
-      clock_(clock),
+      drifting_clock_(clock, speed),
+      clock_(speed == DriftingClock::kNoDrift ? clock : &drifting_clock_),

[pbos-webrtc, 2016/02/10 13:28:14]

Just always use the drifting clock if it's a no-op at kNoDrift?

[danilchap, 2016/02/10 15:50:19]

Simplified. After few failed tries to explain my reasons for using clock
directly realized my reasons are not strong enough.

       tick_(EventTimerWrapper::Create()),
       thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
       file_utility_(new ModuleFileUtility(0)),
@@ -51,7 +55,7 @@ int32_t FakeAudioDevice::Init() {
   if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
     return -1;
 
-  if (!tick_->StartTimer(true, 10))
+  if (!tick_->StartTimer(true, 10 / speed_))
     return -1;
   thread_.Start();
   thread_.SetPriority(rtc::kHighPriority);