Wire up transport sequence number / send time callbacks to webrtc via libjingle.

talk/media/webrtc/webrtcvideoengine2_unittest.cc

 /*
  * libjingle
  * Copyright 2004 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 240 matching lines @@
   for (size_t i = 0; i < extensions.size(); ++i) {
     if (extensions[i].uri == kRtpAbsoluteSenderTimeHeaderExtension) {
       EXPECT_EQ(kRtpAbsoluteSenderTimeHeaderExtensionDefaultId,
                 extensions[i].id);
       return;
     }
   }
   FAIL() << "Absolute Sender Time extension not in header-extension list.";
 }
 
+TEST_F(WebRtcVideoEngine2Test, SupportsTransportSequenceNumberHeaderExtension) {
+  std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
+  ASSERT_FALSE(extensions.empty());
+  for (size_t i = 0; i < extensions.size(); ++i) {
+    if (extensions[i].uri == kRtpTransportSequenceNumberHeaderExtension) {
+      EXPECT_EQ(kRtpTransportSequenceNumberHeaderExtensionDefaultId,
+                extensions[i].id);
+      return;
+    }
+  }
+  FAIL() << "Transport sequence number extension not in header-extension list.";
+}
+
 TEST_F(WebRtcVideoEngine2Test, SupportsVideoRotationHeaderExtension) {
   std::vector<RtpHeaderExtension> extensions = engine_.rtp_header_extensions();
   ASSERT_FALSE(extensions.empty());
   for (size_t i = 0; i < extensions.size(); ++i) {
     if (extensions[i].uri == kRtpVideoRotationHeaderExtension) {
       EXPECT_EQ(kRtpVideoRotationHeaderExtensionDefaultId, extensions[i].id);
       return;
     }
   }
   FAIL() << "Video Rotation extension not in header-extension list.";
@@ skipping 898 matching lines @@
 // Test support for absolute send time header extension.
 TEST_F(WebRtcVideoChannel2Test, SendAbsoluteSendTimeHeaderExtensions) {
   TestSetSendRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension,
                                  webrtc::RtpExtension::kAbsSendTime);
 }
 TEST_F(WebRtcVideoChannel2Test, RecvAbsoluteSendTimeHeaderExtensions) {
   TestSetRecvRtpHeaderExtensions(kRtpAbsoluteSenderTimeHeaderExtension,
                                  webrtc::RtpExtension::kAbsSendTime);
 }
 
+// Test support for transport sequence number header extension.
+TEST_F(WebRtcVideoChannel2Test, SendTransportSequenceNumberHeaderExtensions) {
+  TestSetSendRtpHeaderExtensions(
+      kRtpTransportSequenceNumberHeaderExtension,
+      webrtc::RtpExtension::kTransportSequenceNumber);
+}
+TEST_F(WebRtcVideoChannel2Test, RecvTransportSequenceNumberHeaderExtensions) {
+  TestSetRecvRtpHeaderExtensions(
+      kRtpTransportSequenceNumberHeaderExtension,
+      webrtc::RtpExtension::kTransportSequenceNumber);
+}
+
 // Test support for video rotation header extension.
 TEST_F(WebRtcVideoChannel2Test, SendVideoRotationHeaderExtensions) {
   TestSetSendRtpHeaderExtensions(kRtpVideoRotationHeaderExtension,
                                  webrtc::RtpExtension::kVideoRotation);
 }
 TEST_F(WebRtcVideoChannel2Test, RecvVideoRotationHeaderExtensions) {
   TestSetRecvRtpHeaderExtensions(kRtpVideoRotationHeaderExtension,
                                  webrtc::RtpExtension::kVideoRotation);
 }
 
@@ skipping 1111 matching lines @@
   // Verify previous value is not modified if dscp option is not set.
   cricket::VideoSendParameters parameters1 = send_parameters_;
   EXPECT_TRUE(channel_->SetSendParameters(parameters1));
   EXPECT_EQ(rtc::DSCP_AF41, network_interface->dscp());
   parameters1.options.dscp.Set(false);
   EXPECT_TRUE(channel_->SetSendParameters(parameters1));
   EXPECT_EQ(rtc::DSCP_DEFAULT, network_interface->dscp());
   channel_->SetInterface(NULL);
 }
 
+TEST_F(WebRtcVideoChannel2Test, OnSentPacketPropagatesToCall) {
+  webrtc::SentPacket packet = fake_call_->last_packet_sent();
+  EXPECT_EQ(-1, packet.transport_sequence_number);
+  EXPECT_EQ(-1, packet.send_time_ms);
+
+  rtc::SentPacket rtc_sent(1, 1234);
+  channel_->OnPacketSent(rtc_sent);
+  packet = fake_call_->last_packet_sent();
+  EXPECT_EQ(rtc_sent.transport_sequence_number,
+            packet.transport_sequence_number);
+  EXPECT_EQ(rtc_sent.send_time_ms, packet.send_time_ms);
+
+  rtc::SentPacket rtc_sent2(2, 1244);
+  channel_->OnPacketSent(rtc_sent2);
+  packet = fake_call_->last_packet_sent();
+  EXPECT_EQ(rtc_sent2.transport_sequence_number,
+            packet.transport_sequence_number);
+  EXPECT_EQ(rtc_sent2.send_time_ms, packet.send_time_ms);
+}
+
 TEST_F(WebRtcVideoChannel2Test, OnReadyToSendSignalsNetworkState) {
   EXPECT_EQ(webrtc::kNetworkUp, fake_call_->GetNetworkState());
 
   channel_->OnReadyToSend(false);
   EXPECT_EQ(webrtc::kNetworkDown, fake_call_->GetNetworkState());
 
   channel_->OnReadyToSend(true);
   EXPECT_EQ(webrtc::kNetworkUp, fake_call_->GetNetworkState());
 }
 
@@ skipping 881 matching lines @@
 // Ensures that the correct settings are applied to the codec when two temporal
 // layer screencasting is enabled, and that the correct simulcast settings are
 // reapplied when disabling screencasting.
 TEST_F(WebRtcVideoChannel2SimulcastTest,
        DISABLED_TwoTemporalLayerScreencastSettings) {
   // TODO(pbos): Implement.
   FAIL() << "Not implemented.";
 }
 
 }  // namespace cricket