Implement timing frames.

webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc

Index: webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
diff --git a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
index d44cf683dc03e4460506f9698b1402ead4086636..f35680b2d1888df392391e575980b70c4076bc86 100644
--- a/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
+++ b/webrtc/modules/rtp_rtcp/source/rtp_sender_unittest.cc
@@ -38,6 +38,7 @@ namespace {
 const int kTransmissionTimeOffsetExtensionId = 1;
 const int kAbsoluteSendTimeExtensionId = 14;
 const int kTransportSequenceNumberExtensionId = 13;
+const int kVideoTimingExtensionId = 12;
 const int kPayload = 100;
 const int kRtxPayload = 98;
 const uint32_t kTimestamp = 10;
@@ -74,6 +75,8 @@ class LoopbackTransportTest : public webrtc::Transport {
                                    kVideoRotationExtensionId);
     receivers_extensions_.Register(kRtpExtensionAudioLevel,
                                    kAudioLevelExtensionId);
+    receivers_extensions_.Register(kRtpExtensionVideoTiming,
+                                   kVideoTimingExtensionId);
   }
 
   bool SendRtp(const uint8_t* data,
@@ -347,6 +350,7 @@ TEST_P(RtpSenderTestWithoutPacer, AssignSequenceNumberSetPaddingTimestamps) {
   EXPECT_EQ(kTimestamp, transport_.last_sent_packet().Timestamp());
 }
 
+
 TEST_P(RtpSenderTestWithoutPacer,
        TransportFeedbackObserverGetsCorrectByteCount) {
   constexpr int kRtpOverheadBytesPerPacket = 12 + 8;
@@ -460,6 +464,58 @@ TEST_P(RtpSenderTest, SendsPacketsWithTransportSequenceNumber) {
   EXPECT_EQ(transport_.last_packet_id_, transport_seq_no);
 }
 
+TEST_P(RtpSenderTestWithoutPacer, WritesTimestampToTimingExtension) {
+  rtp_sender_->SetStorePacketsStatus(true, 10);
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
+                     kRtpExtensionVideoTiming,
+                     kVideoTimingExtensionId));
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
+                     kRtpExtensionTransmissionTimeOffset,
+                     kTransmissionTimeOffsetExtensionId));
+  EXPECT_EQ(
+      0, rtp_sender_->RegisterRtpHeaderExtension(kRtpExtensionAbsoluteSendTime,
+                                                 kAbsoluteSendTimeExtensionId));

[sprang_webrtc, 2017/06/13 14:14:14]

why toffset and ast extensions?

[ilnik, 2017/06/13 14:55:44]

Leftover from copypaste and playing around. Removed.

+  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);
+  VideoTiming kVideoTiming = {0u, 0u, 0u, 0u, 0u, 0u, true};

[sprang_webrtc, 2017/06/13 14:14:14]

const

[ilnik, 2017/06/13 14:55:44]

Done.

+  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);
+
+  EXPECT_TRUE(rtp_sender_->SendToNetwork(std::move(packet),
+                                         kAllowRetransmission,
+                                         RtpPacketSender::kNormalPriority));
+  EXPECT_EQ(1, transport_.packets_sent());
+  EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
+
+  transport_.last_sent_packet().GetHeader(&rtp_header);
+  EXPECT_TRUE(rtp_header.extension.hasVideoTiming);
+  EXPECT_EQ(kStoredTimeInMs,
+              rtp_header.extension.videoTiming.pacer_exit_ms_delta);

[sprang_webrtc, 2017/06/13 14:14:14]

nit: indentation?

[ilnik, 2017/06/13 14:55:44]

Done.

+
+  fake_clock_.AdvanceTimeMilliseconds(kStoredTimeInMs);
+  rtp_sender_->TimeToSendPacket(kSsrc, kSeqNum, capture_time_ms, false,
+                                PacedPacketInfo());
+
+  EXPECT_EQ(2, transport_.packets_sent());
+  EXPECT_EQ(packet_size, transport_.last_sent_packet().size());
+
+  transport_.last_sent_packet().GetHeader(&rtp_header);
+  EXPECT_TRUE(rtp_header.extension.hasVideoTiming);
+  EXPECT_EQ(kStoredTimeInMs * 2,
+            rtp_header.extension.videoTiming.pacer_exit_ms_delta);
+}
+
 TEST_P(RtpSenderTest, TrafficSmoothingWithExtensions) {
   EXPECT_CALL(mock_paced_sender_, InsertPacket(RtpPacketSender::kNormalPriority,
                                                kSsrc, kSeqNum, _, _, _));