Add histogram stats for AV sync stream offset:

webrtc/video/vie_sync_module.cc

Index: webrtc/video/vie_sync_module.cc
diff --git a/webrtc/video/vie_sync_module.cc b/webrtc/video/vie_sync_module.cc
index f8376e53d1f6d0bf49b715ad09596bfd7201f556..780f5030629bc5b7f76aca5a5f00b972765e4d54 100644
--- a/webrtc/video/vie_sync_module.cc
+++ b/webrtc/video/vie_sync_module.cc
@@ -10,16 +10,32 @@
 
 #include "webrtc/video/vie_sync_module.h"
 
+#include <algorithm>
+
 #include "webrtc/base/checks.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/trace_event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_receiver.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_rtcp.h"
 #include "webrtc/modules/video_coding/include/video_coding.h"
+#include "webrtc/system_wrappers/include/clock.h"
 #include "webrtc/video/stream_synchronization.h"
+#include "webrtc/video_frame.h"
 #include "webrtc/voice_engine/include/voe_video_sync.h"
 
 namespace webrtc {
+namespace {
+bool RtpTimestampToNtp(uint32_t timestamp,
+                       const RtcpList& rtcp_list,
+                       int64_t* timestamp_ms) {
+  if (rtcp_list.size() != 2)
+    return false;

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

Maybe we should simply move this check into RtpToNtpMs and remove this new
function?

[åsapersson, 2016/03/09 15:44:35]

Done.

+
+  if (!RtpToNtpMs(timestamp, rtcp_list, timestamp_ms))
+    return false;
+
+  return true;
+}
 
 int UpdateMeasurements(StreamSynchronization::Measurements* stream,
                        const RtpRtcp& rtp_rtcp, const RtpReceiver& receiver) {
@@ -47,16 +63,17 @@ int UpdateMeasurements(StreamSynchronization::Measurements* stream,
 
   return 0;
 }
+}  // namespace
 
 ViESyncModule::ViESyncModule(VideoCodingModule* vcm)
     : vcm_(vcm),
+      clock_(Clock::GetRealTimeClock()),
       video_receiver_(NULL),
       video_rtp_rtcp_(NULL),
       voe_channel_id_(-1),
       voe_sync_interface_(NULL),
       last_sync_time_(TickTime::Now()),
-      sync_() {
-}
+      sync_() {}
 
 ViESyncModule::~ViESyncModule() {
 }
@@ -157,4 +174,32 @@ void ViESyncModule::Process() {
   vcm_->SetMinimumPlayoutDelay(target_video_delay_ms);
 }
 
+int64_t ViESyncModule::GetStreamSyncOffsetInMs(const VideoFrame& frame) {
+  rtc::CritScope lock(&data_cs_);
+  if (voe_channel_id_ == -1)
+    return -1;
+
+  uint32_t playout_timestamp = 0;
+  if (voe_sync_interface_->GetPlayoutTimestamp(voe_channel_id_,
+                                               playout_timestamp) != 0) {
+    return -1;
+  }
+  int64_t latest_audio_ntp;
+  if (!RtpTimestampToNtp(playout_timestamp, audio_measurement_.rtcp,
+                         &latest_audio_ntp)) {
+    return -1;
+  }
+  int64_t latest_video_ntp;
+  if (!RtpTimestampToNtp(frame.timestamp(), video_measurement_.rtcp,
+                         &latest_video_ntp)) {
+    return -1;
+  }
+  int64_t time_to_render_ms =
+      frame.render_time_ms() - clock_->TimeInMilliseconds();
+  if (time_to_render_ms > 0)
+    latest_video_ntp += time_to_render_ms;
+
+  return std::abs(latest_audio_ntp - latest_video_ntp);
+}
+
 }  // namespace webrtc