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

talk/media/webrtc/webrtcvideoengine2.cc

 /*
  * libjingle
  * Copyright 2014 Google Inc.
  *
  * Redistribution and use in source and binary forms, with or without
  * modification, are permitted provided that the following conditions are met:
  *
  *  1. Redistributions of source code must retain the above copyright notice,
  *     this list of conditions and the following disclaimer.
  *  2. Redistributions in binary form must reproduce the above copyright notice,
@@ skipping 28 matching lines @@
 #include "talk/media/webrtc/webrtcvideoencoderfactory.h"
 #include "talk/media/webrtc/webrtcvideoframe.h"
 #include "talk/media/webrtc/webrtcvoiceengine.h"
 #include "webrtc/base/buffer.h"
 #include "webrtc/base/logging.h"
 #include "webrtc/base/stringutils.h"
 #include "webrtc/call.h"
 #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"
 
 #define UNIMPLEMENTED                                                 \
   LOG(LS_ERROR) << "Call to unimplemented function " << __FUNCTION__; \
   RTC_NOTREACHED()
 
 namespace cricket {
 namespace {
@@ skipping 1597 matching lines @@
       external_encoder_factory_(external_encoder_factory),
       stream_(NULL),
       parameters_(webrtc::VideoSendStream::Config(),
                   options,
                   max_bitrate_bps,
                   codec_settings),
       allocated_encoder_(NULL, webrtc::kVideoCodecUnknown, false),
       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);
   sp.GetFidSsrcs(parameters_.config.rtp.ssrcs,
                  &parameters_.config.rtp.rtx.ssrcs);
   parameters_.config.rtp.c_name = sp.cname;
   parameters_.config.rtp.extensions = rtp_extensions;
 
   VideoCodecSettings params;
   if (codec_settings.Get(&params)) {
@@ skipping 43 matching lines @@
     DCHECK(format_.height == 0);
     LOG(LS_VERBOSE) << "VideoFormat 0x0 set, Dropping frame.";
     return;
   }
   if (muted_) {
     // Create a black frame to transmit instead.
     CreateBlackFrame(&video_frame,
                      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());
 
   LOG(LS_VERBOSE) << "IncomingCapturedFrame: " << video_frame.width() << "x"
                   << video_frame.height() << " -> (codec) "
                   << parameters_.encoder_config.streams.back().width << "x"
                   << parameters_.encoder_config.streams.back().height;
   stream_->Input()->IncomingCapturedFrame(video_frame);
 }
 
 bool WebRtcVideoChannel2::WebRtcVideoSendStream::SetCapturer(
     VideoCapturer* capturer) {
   TRACE_EVENT0("webrtc", "WebRtcVideoSendStream::SetCapturer");
   if (!DisconnectCapturer() && capturer == NULL) {
     return false;
   }
 
   {
     rtc::CritScope cs(&lock_);
 
     if (capturer == NULL) {
       if (stream_ != NULL) {
         LOG(LS_VERBOSE) << "Disabling capturer, sending black frame.";
         webrtc::VideoFrame black_frame;
 
         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);
       }
 
       capturer_ = NULL;
       return true;
     }
 
     capturer_ = capturer;
   }
   // Lock cannot be held while connecting the capturer to prevent lock-order
@@ skipping 827 matching lines @@
       video_codecs[i].rtx_payload_type = rtx_mapping[video_codecs[i].codec.id];
     }
   }
 
   return video_codecs;
 }
 
 }  // namespace cricket
 
 #endif  // HAVE_WEBRTC_VIDEO