Add transport sequence number on the non-pacer path of the rtp sender.

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 58 matching lines @@
 uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
   return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
 }
 
 class LoopbackTransportTest : public webrtc::Transport {
  public:
   LoopbackTransportTest()
       : packets_sent_(0),
         last_sent_packet_len_(0),
         total_bytes_sent_(0),
-        last_sent_packet_(nullptr) {}
+        last_sent_packet_(nullptr),
+        last_packet_id_(-1) {}
 
   ~LoopbackTransportTest() {
     STLDeleteContainerPointers(sent_packets_.begin(), sent_packets_.end());
   }
   bool SendRtp(const uint8_t* data,
                size_t len,
                const PacketOptions& options) override {
     packets_sent_++;
     rtc::Buffer* buffer =
         new rtc::Buffer(reinterpret_cast<const uint8_t*>(data), len);
     last_sent_packet_ = buffer->data();
     last_sent_packet_len_ = len;
+    last_packet_id_ = options.packet_id;
     total_bytes_sent_ += len;
     sent_packets_.push_back(buffer);
     return true;
   }
   bool SendRtcp(const uint8_t* data, size_t len) override { return false; }
   int packets_sent_;
   size_t last_sent_packet_len_;
   size_t total_bytes_sent_;
   uint8_t* last_sent_packet_;
+  int last_packet_id_;
   std::vector<rtc::Buffer*> sent_packets_;
 };
 
 }  // namespace
 
 class MockRtpPacketSender : public RtpPacketSender {
  public:
   MockRtpPacketSender() {}
   virtual ~MockRtpPacketSender() {}
 
   MOCK_METHOD6(InsertPacket,
                void(Priority priority,
                     uint32_t ssrc,
                     uint16_t sequence_number,
                     int64_t capture_time_ms,
                     size_t bytes,
                     bool retransmission));
 };
 
+class MockTransportSequenceNumberAllocator
+    : public TransportSequenceNumberAllocator {
+ public:
+  MOCK_METHOD0(AllocateSequenceNumber, uint16_t());
+};
+
 class RtpSenderTest : public ::testing::Test {
  protected:
   RtpSenderTest()
       : fake_clock_(kStartTime),
         mock_rtc_event_log_(),
         mock_paced_sender_(),
         rtp_sender_(),
         payload_(kPayload),
         transport_(),
         kMarkerBit(true) {
   }
 
   void SetUp() override { SetUpRtpSender(true); }
 
   void SetUpRtpSender(bool pacer) {
     rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport_, nullptr,
                                     pacer ? &mock_paced_sender_ : nullptr,
-                                    nullptr, nullptr, nullptr, nullptr, nullptr,
-                                    &mock_rtc_event_log_));
+                                    &seq_num_allocator_, nullptr, nullptr,
+                                    nullptr, nullptr, &mock_rtc_event_log_));
     rtp_sender_->SetSequenceNumber(kSeqNum);
   }
 
   SimulatedClock fake_clock_;
   MockRtcEventLog mock_rtc_event_log_;
   MockRtpPacketSender mock_paced_sender_;
+  MockTransportSequenceNumberAllocator seq_num_allocator_;
   rtc::scoped_ptr<RTPSender> rtp_sender_;
   int payload_;
   LoopbackTransportTest transport_;
   const bool kMarkerBit;
   uint8_t packet_[kMaxPacketLength];
 
   void VerifyRTPHeaderCommon(const RTPHeader& rtp_header) {
     VerifyRTPHeaderCommon(rtp_header, kMarkerBit, 0);
   }
 
@@ skipping 302 matching lines @@
   webrtc::RTPHeader rtp_header2;
   const bool valid_rtp_header2 = rtp_parser.Parse(&rtp_header2, nullptr);
 
   ASSERT_TRUE(valid_rtp_header2);
   VerifyRTPHeaderCommon(rtp_header2);
   EXPECT_EQ(length, rtp_header2.headerLength);
   EXPECT_FALSE(rtp_header2.extension.hasAbsoluteSendTime);
   EXPECT_EQ(0u, rtp_header2.extension.absoluteSendTime);
 }
 
+TEST_F(RtpSenderTestWithoutPacer, SendsPacketsWithTransportSequenceNumber) {
+  // Ignore rtc event calls.
+  EXPECT_CALL(mock_rtc_event_log_,
+              LogRtpHeader(PacketDirection::kOutgoingPacket, _, _, _));
+
+  EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
+                   kRtpExtensionTransportSequenceNumber,
+                   kTransportSequenceNumberExtensionId));
+
+  char payload_name[RTP_PAYLOAD_NAME_SIZE] = "GENERIC";
+  const uint8_t payload_type = 127;
+  ASSERT_EQ(0, rtp_sender_->RegisterPayload(payload_name, payload_type, 90000,
+                                            0, 1500));
+  uint8_t payload[] = {47, 11, 32, 93, 89};

[sprang_webrtc, 2016/01/27 11:32:17]

There's no special meaning to these values, right?

[stefan-webrtc, 2016/01/27 11:39:41]

Nope :)

+
+  EXPECT_CALL(seq_num_allocator_, AllocateSequenceNumber())
+      .WillOnce(testing::Return(17));

[sprang_webrtc, 2016/01/27 11:32:17]

Name 17 and check it at the end.

[stefan-webrtc, 2016/01/27 11:39:41]

Done.

+  ASSERT_EQ(
+      0, rtp_sender_->SendOutgoingData(kVideoFrameKey, payload_type, 1234, 4321,

[sprang_webrtc, 2016/01/27 11:32:17]

Nit: Perhaps name or comment on what 1234 and 4321 is?

[stefan-webrtc, 2016/01/27 11:39:41]

Done.

+                                       payload, sizeof(payload), nullptr));
+
+  RtpUtility::RtpHeaderParser rtp_parser(transport_.last_sent_packet_,
+                                         transport_.last_sent_packet_len_);
+  webrtc::RTPHeader rtp_header;
+  RtpHeaderExtensionMap map;
+  map.Register(kRtpExtensionTransportSequenceNumber,
+               kTransportSequenceNumberExtensionId);
+  EXPECT_TRUE(rtp_parser.Parse(&rtp_header, &map));
+  EXPECT_TRUE(rtp_header.extension.hasTransportSequenceNumber);
+  EXPECT_EQ(transport_.last_packet_id_,
+            rtp_header.extension.transportSequenceNumber);
+}
+
 // Test CVO header extension is only set when marker bit is true.
 TEST_F(RtpSenderTestWithoutPacer, BuildRTPPacketWithVideoRotation_MarkerBit) {
   rtp_sender_->SetVideoRotation(kRotation);
   EXPECT_EQ(0, rtp_sender_->RegisterRtpHeaderExtension(
                    kRtpExtensionVideoRotation, kVideoRotationExtensionId));
   EXPECT_TRUE(rtp_sender_->ActivateCVORtpHeaderExtension());
 
   RtpHeaderExtensionMap map;
   map.Register(kRtpExtensionVideoRotation, kVideoRotationExtensionId);
 
@@ skipping 1000 matching lines @@
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[0]->data()),
       transport_.sent_packets_[0]->size(), true, &map, kSeqNum, hdr.rotation);
 
   // Verify that this packet does have CVO byte.
   VerifyCVOPacket(
       reinterpret_cast<uint8_t*>(transport_.sent_packets_[1]->data()),
       transport_.sent_packets_[1]->size(), true, &map, kSeqNum + 1,
       hdr.rotation);
 }
 }  // namespace webrtc