Configured VCMTiming with sender defining delay times.

BUG=webrtc:7590

Review-Url: https://codereview.webrtc.org/2870823003
Cr-Original-Commit-Position: refs/heads/master@{#18086}
Review-Url: https://codereview.webrtc.org/2890943002 .
Cr-Commit-Position: refs/branch-heads/59@{#10}
Cr-Branched-From: 10d095d-refs/heads/master@{#17657}

Author: SergeyUlanov

webrtc/modules/video_coding/encoded_frame.h

@@ -46,6 +46,11 @@ class VCMEncodedFrame : protected EncodedImage {
     _encodedWidth = width;
     _encodedHeight = height;
   }
+
+  void SetPlayoutDelay(PlayoutDelay playout_delay) {
+    playout_delay_ = playout_delay;
+  }
+
   /**
   *   Get the encoded image
   */

webrtc/modules/video_coding/frame_buffer2.cc

@@ -43,6 +43,7 @@ FrameBuffer::FrameBuffer(Clock* clock,
       jitter_estimator_(jitter_estimator),
       timing_(timing),
       inter_frame_delay_(clock_->TimeInMilliseconds()),
+      last_decoded_frame_timestamp_(0),
       last_decoded_frame_it_(frames_.end()),
       last_continuous_frame_it_(frames_.end()),
       num_frames_history_(0),
@@ -180,6 +181,16 @@ void FrameBuffer::Stop() {
   new_continuous_frame_event_.Set();
 }
 
+void FrameBuffer::UpdatePlayoutDelays(const FrameObject& 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);
+}
+
 int FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
   TRACE_EVENT0("webrtc", "FrameBuffer::InsertFrame");
   RTC_DCHECK(frame);
@@ -256,7 +267,7 @@ int FrameBuffer::InsertFrame(std::unique_ptr<FrameObject> frame) {
 
   if (!UpdateFrameInfoWithIncomingFrame(*frame, info))
     return last_continuous_picture_id;
-
+  UpdatePlayoutDelays(*frame);
   info->second.frame = std::move(frame);
   ++num_frames_buffered_;
 

webrtc/modules/video_coding/frame_buffer2.h

@@ -120,6 +120,11 @@ class FrameBuffer {
 
   using FrameMap = std::map<FrameKey, FrameInfo>;
 
+  // Updates the minimal and maximal playout delays
+  // depending on the frame.
+  void UpdatePlayoutDelays(const FrameObject& frame)
+      EXCLUSIVE_LOCKS_REQUIRED(crit_);
+
   // Update all directly dependent and indirectly dependent frames and mark
   // them as continuous if all their references has been fulfilled.
   void PropagateContinuity(FrameMap::iterator start)

webrtc/modules/video_coding/frame_buffer2_unittest.cc

@@ -260,6 +260,15 @@ TEST_F(TestFrameBuffer2, OneSuperFrame) {
   CheckFrame(1, pid, 1);
 }
 
+TEST_F(TestFrameBuffer2, SetPlayoutDelay) {
+  const PlayoutDelay kPlayoutDelayMs = {123, 321};
+  std::unique_ptr<FrameObjectFake> test_frame(new FrameObjectFake());
+  test_frame->SetPlayoutDelay(kPlayoutDelayMs);
+  buffer_.InsertFrame(std::move(test_frame));
+  EXPECT_EQ(kPlayoutDelayMs.min_ms, timing_.min_playout_delay());
+  EXPECT_EQ(kPlayoutDelayMs.max_ms, timing_.max_playout_delay());
+}
+
 // Flaky test, see bugs.webrtc.org/7068.
 TEST_F(TestFrameBuffer2, DISABLED_OneUnorderedSuperFrame) {
   uint16_t pid = Rand();

webrtc/modules/video_coding/frame_object.cc

@@ -48,6 +48,10 @@ RtpFrameObject::RtpFrameObject(PacketBuffer* packet_buffer,
   _timeStamp = first_packet->timestamp;
   ntp_time_ms_ = first_packet->ntp_time_ms_;
 
+  // Setting frame's playout delays to the same values
+  // as of the first packet's.
+  SetPlayoutDelay(first_packet->video_header.playout_delay);
+
   // Since FFmpeg use an optimized bitstream reader that reads in chunks of
   // 32/64 bits we have to add at least that much padding to the buffer
   // to make sure the decoder doesn't read out of bounds.