Propagate base minimum delay from video jitter buffer to webrtc/api.

On api level two methods were added to api/media_stream_interface.cc on VideoSourceInterface,
GetLatency and SetLatency. Latency is measured in seconds, delay in milliseconds but both describes
the same concept.


Bug: webrtc:10287
Change-Id: Ib8dc62a4d73f63fab7e10b82c716096ee6199957
Reviewed-on: https://webrtc-review.googlesource.com/c/123482
Commit-Queue: Ruslan Burakov <kuddai@google.com>
Reviewed-by: Stefan Holmer <stefan@webrtc.org>
Reviewed-by: Philip Eliasson <philipel@webrtc.org>
Reviewed-by: Steve Anton <steveanton@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#26877}
diff --git a/modules/video_coding/frame_buffer2.cc b/modules/video_coding/frame_buffer2.cc
index 6401117..e4def86 100644
--- a/modules/video_coding/frame_buffer2.cc
+++ b/modules/video_coding/frame_buffer2.cc
@@ -347,19 +347,6 @@
   return true;
 }
 
-void FrameBuffer::UpdatePlayoutDelays(const EncodedFrame& frame) {
-  TRACE_EVENT0("webrtc", "FrameBuffer::UpdatePlayoutDelays");
-  PlayoutDelay playout_delay = frame.EncodedImage().playout_delay_;
-  if (playout_delay.min_ms >= 0)
-    timing_->set_min_playout_delay(playout_delay.min_ms);
-
-  if (playout_delay.max_ms >= 0)
-    timing_->set_max_playout_delay(playout_delay.max_ms);
-
-  if (!frame.delayed_by_retransmission())
-    timing_->IncomingTimestamp(frame.Timestamp(), frame.ReceivedTime());
-}
-
 int64_t FrameBuffer::InsertFrame(std::unique_ptr<EncodedFrame> frame) {
   TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame");
   RTC_DCHECK(frame);
@@ -449,7 +436,9 @@
 
   if (!UpdateFrameInfoWithIncomingFrame(*frame, info))
     return last_continuous_picture_id;
-  UpdatePlayoutDelays(*frame);
+
+  if (!frame->delayed_by_retransmission())
+    timing_->IncomingTimestamp(frame->Timestamp(), frame->ReceivedTime());
 
   info->second.frame = std::move(frame);