Implement RTP keepalive in native stack.

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 39 matching lines @@
 const uint8_t kAudioLevelExtensionId = 9;
 const int kAudioPayload = 103;
 const uint64_t kStartTime = 123456789;
 const size_t kMaxPaddingSize = 224u;
 const int kVideoRotationExtensionId = 5;
 const size_t kGenericHeaderLength = 1;
 const uint8_t kPayloadData[] = {47, 11, 32, 93, 89};
 
 using ::testing::_;
 using ::testing::ElementsAreArray;
+using ::testing::Invoke;
 
 uint64_t ConvertMsToAbsSendTime(int64_t time_ms) {
   return (((time_ms << 18) + 500) / 1000) & 0x00ffffff;
 }
 
 class LoopbackTransportTest : public webrtc::Transport {
  public:
   LoopbackTransportTest() : total_bytes_sent_(0), last_packet_id_(-1) {
     receivers_extensions_.Register(kRtpExtensionTransmissionTimeOffset,
                                    kTransmissionTimeOffsetExtensionId);
@@ skipping 1634 matching lines @@
 
   // Requested padding size is too small, will send a larger one.
   const size_t kMinPaddingSize = 50;
   EXPECT_CALL(transport, SendRtp(_, kMinPaddingSize + kRtpHeaderSize, _))
       .WillOnce(testing::Return(true));
   EXPECT_EQ(
       kMinPaddingSize,
       rtp_sender_->TimeToSendPadding(kMinPaddingSize - 5, PacedPacketInfo()));
 }
 
+TEST_P(RtpSenderTest, SendsKeepAlive) {
+  MockTransport transport;
+  rtp_sender_.reset(new RTPSender(false, &fake_clock_, &transport, nullptr,
+                                  nullptr, nullptr, nullptr, nullptr, nullptr,
+                                  nullptr, &mock_rtc_event_log_, nullptr,
+                                  &retransmission_rate_limiter_, nullptr));
+  rtp_sender_->SetSendPayloadType(kPayload);
+  rtp_sender_->SetSequenceNumber(kSeqNum);
+  rtp_sender_->SetTimestampOffset(0);
+  rtp_sender_->SetSSRC(kSsrc);
+
+  const uint8_t kKeepalivePayloadType = 20;
+  RTC_CHECK_NE(kKeepalivePayloadType, kPayload);
+
+  EXPECT_CALL(transport, SendRtp(_, _, _))
+      .WillOnce(
+          Invoke([&kKeepalivePayloadType](const uint8_t* packet, size_t len,
+                                          const PacketOptions& options) {
+            webrtc::RTPHeader rtp_header;
+            RtpUtility::RtpHeaderParser parser(packet, len);
+            EXPECT_TRUE(parser.Parse(&rtp_header, nullptr));
+            EXPECT_TRUE(rtp_header.markerBit);
+            EXPECT_EQ(0U, rtp_header.paddingLength);
+            EXPECT_EQ(kKeepalivePayloadType, rtp_header.payloadType);
+            EXPECT_EQ(kSsrc, rtp_header.ssrc);
+            EXPECT_EQ(0u, len - rtp_header.headerLength);
+            return true;
+          }));
+
+  rtp_sender_->SendKeepAlive(kKeepalivePayloadType);
+}
+
 INSTANTIATE_TEST_CASE_P(WithAndWithoutOverhead,
                         RtpSenderTest,
                         ::testing::Bool());
 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