Added A/V sync tests with drifting clocks.

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.
  */
 #include <algorithm>
 #include <sstream>
 #include <string>
 
 #include "testing/gtest/include/gtest/gtest.h"
 
 #include "webrtc/base/checks.h"
 #include "webrtc/base/scoped_ptr.h"
 #include "webrtc/base/thread_annotations.h"
 #include "webrtc/call.h"
 #include "webrtc/call/transport_adapter.h"
 #include "webrtc/config.h"
 #include "webrtc/modules/audio_coding/include/audio_coding_module.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
 #include "webrtc/modules/rtp_rtcp/source/rtcp_utility.h"
 #include "webrtc/system_wrappers/include/critical_section_wrapper.h"
 #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"
 #include "webrtc/test/fake_encoder.h"
 #include "webrtc/test/frame_generator.h"
 #include "webrtc/test/frame_generator_capturer.h"
 #include "webrtc/test/rtp_rtcp_observer.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/voice_engine/include/voe_base.h"
 #include "webrtc/voice_engine/include/voe_codec.h"
 #include "webrtc/voice_engine/include/voe_network.h"
 #include "webrtc/voice_engine/include/voe_rtp_rtcp.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 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);
 
   void TestMinTransmitBitrate(bool pad_to_min_bitrate);
 
   void TestCaptureNtpTime(const FakeNetworkPipe::Config& net_config,
                           int threshold_ms,
                           int start_time_ms,
                           int run_time_ms);
 };
@@ skipping 122 matching lines @@
 
  private:
   Clock* const clock_;
   const int voe_channel_;
   VoEVideoSync* const voe_sync_;
   SyncRtcpObserver* const audio_observer_;
   const int64_t creation_time_ms_;
   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;
   class AudioPacketReceiver : public PacketReceiver {
    public:
     AudioPacketReceiver(int channel, VoENetwork* voe_network)
         : channel_(channel),
           voe_network_(voe_network),
           parser_(RtpHeaderParser::Create()) {}
     DeliveryStatus DeliverPacket(MediaType media_type,
@@ skipping 20 matching lines @@
 
   VoiceEngine* voice_engine = VoiceEngine::Create();
   VoEBase* voe_base = VoEBase::GetInterface(voice_engine);
   VoECodec* voe_codec = VoECodec::GetInterface(voice_engine);
   VoENetwork* voe_network = VoENetwork::GetInterface(voice_engine);
   VoEVideoSync* voe_sync = VoEVideoSync::GetInterface(voice_engine);
   const std::string audio_filename =
       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));
   int send_channel_id = voe_base->CreateChannel(voe_config);
   int recv_channel_id = voe_base->CreateChannel();
 
   SyncRtcpObserver audio_observer;
 
   AudioState::Config send_audio_state_config;
   send_audio_state_config.voice_engine = voice_engine;
@@ skipping 74 matching lines @@
   if (create_audio_first) {
     audio_receive_stream =
         receiver_call_->CreateAudioReceiveStream(audio_recv_config);
     CreateVideoStreams();
   } else {
     CreateVideoStreams();
     audio_receive_stream =
         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();
 
   fake_audio_device.Start();
   EXPECT_EQ(0, voe_base->StartPlayout(recv_channel_id));
   EXPECT_EQ(0, voe_base->StartReceive(recv_channel_id));
   EXPECT_EQ(0, voe_base->StartSend(send_channel_id));
 
   EXPECT_TRUE(observer.Wait())
       << "Timed out while waiting for audio and video to be synchronized.";
@@ skipping 20 matching lines @@
   voe_codec->Release();
   voe_network->Release();
   voe_sync->Release();
 
   DestroyCalls();
 
   VoiceEngine::Delete(voice_engine);
 }
 
 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,
                                       int threshold_ms,
                                       int start_time_ms,
                                       int run_time_ms) {
   class CaptureNtpTimeObserver : public test::EndToEndTest,
                                  public VideoRenderer {
    public:
     CaptureNtpTimeObserver(const FakeNetworkPipe::Config& net_config,
@@ skipping 362 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc