Let WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame carry the input frame's timestamp to out…

talk/media/webrtc/webrtcvideoengine2.cc

Index: talk/media/webrtc/webrtcvideoengine2.cc
diff --git a/talk/media/webrtc/webrtcvideoengine2.cc b/talk/media/webrtc/webrtcvideoengine2.cc
index 0a2152e5e4c891494f34df4a271acb13aa3ec90f..10fa54d04c2c97fdb3956149ae172103b3c16261 100644
--- a/talk/media/webrtc/webrtcvideoengine2.cc
+++ b/talk/media/webrtc/webrtcvideoengine2.cc
@@ -46,6 +46,7 @@
 #include "webrtc/modules/video_coding/codecs/h264/include/h264.h"
 #include "webrtc/modules/video_coding/codecs/vp8/simulcast_encoder_adapter.h"
 #include "webrtc/system_wrappers/interface/field_trial.h"
+#include "webrtc/system_wrappers/interface/tick_util.h"
 #include "webrtc/system_wrappers/interface/trace_event.h"
 #include "webrtc/video_decoder.h"
 #include "webrtc/video_encoder.h"
@@ -1663,7 +1664,9 @@ WebRtcVideoChannel2::WebRtcVideoSendStream::WebRtcVideoSendStream(
       capturer_(NULL),
       sending_(false),
       muted_(false),
-      old_adapt_changes_(0) {
+      old_adapt_changes_(0),
+      base_timestamp_ms_(0),
+      last_timestamp_ms_(0) {
   parameters_.config.rtp.max_packet_size = kVideoMtu;
 
   sp.GetPrimarySsrcs(&parameters_.config.rtp.ssrcs);
@@ -1727,6 +1730,15 @@ void WebRtcVideoChannel2::WebRtcVideoSendStream::InputFrame(
                      static_cast<int>(frame->GetWidth()),
                      static_cast<int>(frame->GetHeight()));
   }
+
+  // frame->GetTimeStamp() is essentially a delta, align to webrtc time
+  if (base_timestamp_ms_ == 0) {
+    base_timestamp_ms_ = webrtc::TickTime::MillisecondTimestamp() -
+                         frame->GetTimeStamp() / 1000000;
+  }
+
+  last_timestamp_ms_ = base_timestamp_ms_ + frame->GetTimeStamp() / 1000000;
+  video_frame.set_render_time_ms(last_timestamp_ms_);

[pthatcher1, 2015/07/14 17:57:28]

I think this would be more clear if you put "frame->GetTimeStamp() / 1000000" on
its own line and used webrtc/base/timeutils.h instead of
webrtc/system_wrappers/interface/tick_util.h:

// frame->GetTimeStamp() is essentially a delta in nanoseconds
uint64 frame_delta_ms_ = frame->GetTimeStamp() / rtc:: kNumNanosecsPerMillisec;
if (base_timestamp_ms_ == 0) {
  base_timestamp_ms_ = rtc::Time() - frame_delta_ms_;
}

last_timestamp_ms_ = base_timestamp_ms_ + frame_delta_ms_;
video_frame.set_render_time_ms(last_timestamp_ms_);

[qiangchen, 2015/07/14 22:04:26]

Done.

   // Reconfigure codec if necessary.
   SetDimensions(
       video_frame.width(), video_frame.height(), capturer->IsScreencast());
@@ -1755,6 +1767,10 @@ bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
 
         CreateBlackFrame(&black_frame, last_dimensions_.width,
                          last_dimensions_.height);
+
+        // Force this black frame not to be dropped due to timestamp order
+        // check.

[pthatcher1, 2015/07/14 17:57:28]

Can you explain a little more how adding "+ 1" accomplishes what you want here?

[qiangchen, 2015/07/14 22:04:26]

Done. Add 1 could bypass the timestamp check in IncomingCapturedFrame, however,
on the rendering side the frame would still be dropped because it is so close to
previous frame. Add a expected frame length.

+        black_frame.set_render_time_ms(last_timestamp_ms_ + 1);
         stream_->Input()->IncomingCapturedFrame(black_frame);
       }