Added A/V sync tests with drifting clocks.

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 3ee49d94271f6e8f875faea615416aed4e3b6ede..fe4fda65cac36cfb4613cffe8e844753a659d7ab 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -26,6 +26,7 @@
 #include "webrtc/system_wrappers/include/rtp_to_ntp.h"
 #include "webrtc/test/call_test.h"
 #include "webrtc/test/direct_transport.h"
+#include "webrtc/test/drifting_clock.h"
 #include "webrtc/test/encoder_settings.h"
 #include "webrtc/test/fake_audio_device.h"
 #include "webrtc/test/fake_decoder.h"
@@ -45,7 +46,11 @@ namespace webrtc {
 
 class CallPerfTest : public test::CallTest {
  protected:
-  void TestAudioVideoSync(bool fec, bool create_audio_first);
+  void TestAudioVideoSync(bool fec,
+                          bool create_audio_first,
+                          float audio_rtp_drift,

[pbos-webrtc, 2016/02/09 20:42:24]

..drift_factor (and use 1.03 and .97 for instance).

[danilchap, 2016/02/10 12:40:35]

renamed to speed. Using values like 1.3 and .7 now, but hiding it with more
(hopefully) readable names.

+                          float video_rtp_drift,
+                          float video_ntp_drift);
 
   void TestCpuOveruse(LoadObserver::Load tested_load, int encode_delay_ms);
 
@@ -188,7 +193,11 @@ class VideoRtcpAndSyncObserver : public SyncRtcpObserver, public VideoRenderer {
   int64_t first_time_in_sync_;
 };
 
-void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
+void CallPerfTest::TestAudioVideoSync(bool fec,
+                                      bool create_audio_first,
+                                      float audio_rtp_drift,
+                                      float video_rtp_drift,
+                                      float video_ntp_drift) {
   const char* kSyncGroup = "av_sync";
   const uint32_t kAudioSendSsrc = 1234;
   const uint32_t kAudioRecvSsrc = 5678;
@@ -229,7 +238,7 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
       test::ResourcePath("voice_engine/audio_long16", "pcm");
   ASSERT_STRNE("", audio_filename.c_str());
   test::FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(),
-                                          audio_filename);
+                                          audio_filename, audio_rtp_drift);
   EXPECT_EQ(0, voe_base->Init(&fake_audio_device, nullptr));
   Config voe_config;
   voe_config.Set<VoicePacing>(new VoicePacing(true));
@@ -324,7 +333,12 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
         receiver_call_->CreateAudioReceiveStream(audio_recv_config);
   }
 
-  CreateFrameGeneratorCapturer();
+  test::DriftingClock drifting_clock(clock_, video_ntp_drift);
+  if (video_ntp_drift == 0 && video_rtp_drift == 0) {
+    CreateFrameGeneratorCapturer();
+  } else {
+    CreateFrameGeneratorCapturerWithDrift(&drifting_clock, video_rtp_drift);
+  }
 
   Start();
 
@@ -365,15 +379,35 @@ void CallPerfTest::TestAudioVideoSync(bool fec, bool create_audio_first) {
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioCreatedFirst) {
-  TestAudioVideoSync(false, true);
+  TestAudioVideoSync(false, true, 0, 0, 0);
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoCreatedFirst) {
-  TestAudioVideoSync(false, false);
+  TestAudioVideoSync(false, false, 0, 0, 0);
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithFec) {
-  TestAudioVideoSync(true, false);
+  TestAudioVideoSync(true, false, 0, 0, 0);
+}
+
+TEST_F(CallPerfTest, DISABLED_PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {

[pbos-webrtc, 2016/02/09 20:42:24]

put a TODO(danilchap): to enable it here with a reference (link) to the bug

[danilchap, 2016/02/10 12:40:35]

Done.

+  TestAudioVideoSync(false, true, 0, 0, 0.02f);
+}
+
+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioRtpDrift) {

[pbos-webrtc, 2016/02/09 20:42:24]

If we can have more aggressive percentages here (order of +10%) I think that's
better (if we can handle it I want us to keep being able to handle it). Same for
the NTP time.

[danilchap, 2016/02/10 12:40:35]

0.3 means 30%. Abstract constants replaced with something readable.

+  TestAudioVideoSync(false, true, 0.3f, 0, 0);
+}
+
+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoRtpDrift) {
+  TestAudioVideoSync(false, true, 0, 0.3f, 0);
+}
+
+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
+  TestAudioVideoSync(false, true, -0.3f, 0.3f, 0);
+}
+
+TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoFasterThanAudioDrift) {
+  TestAudioVideoSync(false, true, 0.3f, -0.3f, 0);
 }
 
 void CallPerfTest::TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,