Report timing frames info in GetStats.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

 /*
  *  Copyright (c) 2012 The WebRTC project authors. All Rights Reserved.
  *
  *  Use of this source code is governed by a BSD-style license
  *  that can be found in the LICENSE file in the root of the source
  *  tree. An additional intellectual property rights grant can be found
  *  in the file PATENTS.  All contributing project authors may
  *  be found in the AUTHORS file in the root of the source tree.
  */
 
@@ skipping 455 matching lines @@
 TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
   rtp_sender_->SetStorePacketsStatus(true, 10);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoTiming, kVideoTimingExtensionId));
   int64_t capture_time_ms = fake_clock_.TimeInMilliseconds();
   auto packet = rtp_sender_->AllocatePacket();
   packet->SetPayloadType(kPayload);
   packet->SetMarker(true);
   packet->SetTimestamp(kTimestamp);
   packet->set_capture_time_ms(capture_time_ms);
-  const VideoTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
+  const VideoSendTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};
   packet->SetExtension<VideoTimingExtension>(kVideoTiming);
   EXPECT_TRUE(rtp_sender_->AssignSequenceNumber(packet.get()));
   size_t packet_size = packet->size();
   webrtc::RTPHeader rtp_header;
 
   packet->GetHeader(&rtp_header);
 
   const int kStoredTimeInMs = 100;
   fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
 
@@ skipping 1079 matching lines @@
 
   RTPVideoHeader hdr = {0};
   hdr.video_timing.is_timing_frame = true;
   hdr.video_timing.encode_start_delta_ms = kEncodeStartDeltaMs;
   hdr.video_timing.encode_finish_delta_ms = kEncodeFinishDeltaMs;
 
   fake_clock_.AdvanceTimeMilliseconds(kPacketizationTimeMs);
   rtp_sender_video_->SendVideo(kRtpVideoGeneric, kVideoFrameKey, kPayload,
                                kTimestamp, kCaptureTimestamp, kFrame,
                                sizeof(kFrame), nullptr, &hdr);
-  VideoTiming timing;
+  VideoSendTiming timing;
   EXPECT_TRUE(transport_.last_sent_packet().GetExtension<VideoTimingExtension>(
       &timing));
   EXPECT_EQ(kPacketizationTimeMs, timing.packetization_finish_delta_ms);
   EXPECT_EQ(kEncodeStartDeltaMs, timing.encode_start_delta_ms);
   EXPECT_EQ(kEncodeFinishDeltaMs, timing.encode_finish_delta_ms);
 }
 
 TEST_P(RtpSenderVideoTest, DeltaFrameHasCVOWhenChanged) {
   uint8_t kFrame[kMaxPacketLength];
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
@@ skipping 125 matching lines @@
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderTestWithoutPacer,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderVideoTest,
                         ::testing::Bool());
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderAudioTest,
                         ::testing::Bool());
 }  // namespace webrtc