Add histogram stats for AV sync stream offset:

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>
@@ skipping 15 matching lines @@
 #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/histogram.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"
 
 using webrtc::test::DriftingClock;
@@ skipping 109 matching lines @@
     if (voe_sync_->GetPlayoutTimestamp(voe_channel_, playout_timestamp) != 0)
       return;
     int64_t latest_audio_ntp =
         audio_observer_->RtpTimestampToNtp(playout_timestamp);
     int64_t latest_video_ntp = RtpTimestampToNtp(video_frame.timestamp());
     if (latest_audio_ntp < 0 || latest_video_ntp < 0)
       return;
     int time_until_render_ms =
         std::max(0, static_cast<int>(video_frame.render_time_ms() - now_ms));
     latest_video_ntp += time_until_render_ms;
     int64_t stream_offset = latest_audio_ntp - latest_video_ntp;

[stefan-webrtc, 2016/03/04 14:32:17]

If you also wire this new metric up to GetStats() on VideoReceiveStream we
should be able to call GetStats() here instead of accessing the VoEVideoSync
interface which should be hidden.

[åsapersson, 2016/03/09 15:44:35]

Done.

     std::stringstream ss;
     ss << stream_offset;
     webrtc::test::PrintResult("stream_offset",
                               "",
                               "synchronization",
                               ss.str(),
                               "ms",
                               false);
     int64_t time_since_creation = now_ms - creation_time_ms_;
     // During the first couple of seconds audio and video can falsely be
@@ skipping 55 matching lines @@
       }
       return ret == 0 ? DELIVERY_OK : DELIVERY_PACKET_ERROR;
     }
 
    private:
     int channel_;
     VoENetwork* voe_network_;
     rtc::scoped_ptr<RtpHeaderParser> parser_;
   };
 
+  test::ClearHistograms();
   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());
   FakeAudioDevice fake_audio_device(Clock::GetRealTimeClock(), audio_filename,
                                     audio_rtp_speed);
@@ skipping 123 matching lines @@
   voe_base->DeleteChannel(send_channel_id);
   voe_base->DeleteChannel(recv_channel_id);
   voe_base->Release();
   voe_codec->Release();
   voe_network->Release();
   voe_sync->Release();
 
   DestroyCalls();
 
   VoiceEngine::Delete(voice_engine);
+
+  EXPECT_EQ(1, test::NumHistogramSamples("WebRTC.Video.AVSyncOffsetInMs"));

[stefan-webrtc, 2016/03/04 14:32:17]

Can we also check that the sync offset is reasonable? Or simply wire up the sync
offset to GetStats() and use above, that way we will have a rough idea if it's
correct or not.

[åsapersson, 2016/03/09 15:44:35]

Wired up to and tested through GetStats().

[stefan-webrtc, 2016/03/09 15:59:30]

Thanks a lot!

 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
   TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
                      DriftingClock::PercentsFaster(10.0f),
                      DriftingClock::kNoDrift, DriftingClock::kNoDrift);
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithAudioFasterThanVideoDrift) {
   TestAudioVideoSync(FecMode::kOff, CreateOrder::kAudioFirst,
@@ skipping 382 matching lines @@
     int encoder_inits_;
     uint32_t last_set_bitrate_;
     VideoSendStream* send_stream_;
     VideoEncoderConfig encoder_config_;
   } test;
 
   RunBaseTest(&test);
 }
 
 }  // namespace webrtc