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..e1454b0f5198935d763000a1ef4b88bf584f8263 100644
--- a/webrtc/test/fake_audio_device.cc
+++ b/webrtc/test/fake_audio_device.cc
@@ -13,6 +13,7 @@
 #include <algorithm>
 
 #include "testing/gtest/include/gtest/gtest.h"
+#include "webrtc/base/checks.h"
 #include "webrtc/base/platform_thread.h"
 #include "webrtc/modules/media_file/media_file_utility.h"
 #include "webrtc/system_wrappers/include/clock.h"
@@ -22,13 +23,17 @@
 namespace webrtc {
 namespace test {
 
-FakeAudioDevice::FakeAudioDevice(Clock* clock, const std::string& filename)
+FakeAudioDevice::FakeAudioDevice(Clock* clock,
+                                 const std::string& filename,
+                                 float drift)
     : audio_callback_(NULL),
       capturing_(false),
       captured_audio_(),
       playout_buffer_(),
+      drift_(drift),
       last_playout_ms_(-1),
-      clock_(clock),
+      drifting_clock_(clock, drift),
+      clock_(drift == 0 ? clock : &drifting_clock_),
       tick_(EventTimerWrapper::Create()),
       thread_(FakeAudioDevice::Run, this, "FakeAudioDevice"),
       file_utility_(new ModuleFileUtility(0)),
@@ -51,7 +56,7 @@ int32_t FakeAudioDevice::Init() {
   if (file_utility_->InitPCMReading(*input_stream_.get()) != 0)
     return -1;
 
-  if (!tick_->StartTimer(true, 10))
+  if (!tick_->StartTimer(true, 10 * (1. + drift_)))
     return -1;
   thread_.Start();
   thread_.SetPriority(rtc::kHighPriority);