Implement RTP keepalive in native stack.

webrtc/modules/rtp_rtcp/source/rtp_rtcp_impl_unittest.cc

 /*
  *  Copyright (c) 2013 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 35 matching lines @@
 
   void OnRttUpdate(int64_t rtt_ms) override { rtt_ms_ = rtt_ms; }
   int64_t LastProcessedRtt() const override { return rtt_ms_; }
   int64_t rtt_ms_;
 };
 
 class SendTransport : public Transport,
                       public RtpData {
  public:
   SendTransport()
-      : receiver_(NULL),
-        clock_(NULL),
+      : receiver_(nullptr),
+        clock_(nullptr),
         delay_ms_(0),
-        rtp_packets_sent_(0) {
-  }
+        rtp_packets_sent_(0),
+        keepalive_payload_type_(0),
+        num_keepalive_sent_(0) {}
 
   void SetRtpRtcpModule(ModuleRtpRtcpImpl* receiver) {
     receiver_ = receiver;
   }
   void SimulateNetworkDelay(int64_t delay_ms, SimulatedClock* clock) {
     clock_ = clock;
     delay_ms_ = delay_ms;
   }
   bool SendRtp(const uint8_t* data,
                size_t len,
                const PacketOptions& options) override {
     RTPHeader header;
     std::unique_ptr<RtpHeaderParser> parser(RtpHeaderParser::Create());
     EXPECT_TRUE(parser->Parse(static_cast<const uint8_t*>(data), len, &header));
     ++rtp_packets_sent_;
+    if (header.payloadType == keepalive_payload_type_)
+      ++num_keepalive_sent_;
     last_rtp_header_ = header;
     return true;
   }
   bool SendRtcp(const uint8_t* data, size_t len) override {
     test::RtcpPacketParser parser;
     parser.Parse(data, len);
     last_nack_list_ = parser.nack()->packet_ids();
 
     if (clock_) {
       clock_->AdvanceTimeMilliseconds(delay_ms_);
     }
     EXPECT_TRUE(receiver_);
     EXPECT_EQ(0, receiver_->IncomingRtcpPacket(data, len));
     return true;
   }
   int32_t OnReceivedPayloadData(const uint8_t* payload_data,
                                 size_t payload_size,
                                 const WebRtcRTPHeader* rtp_header) override {
     return 0;
   }
+  void SetKeepalivePayloadType(uint8_t payload_type) {
+    keepalive_payload_type_ = payload_type;
+  }
+  size_t NumKeepaliveSent() { return num_keepalive_sent_; }
   ModuleRtpRtcpImpl* receiver_;
   SimulatedClock* clock_;
   int64_t delay_ms_;
   int rtp_packets_sent_;
   RTPHeader last_rtp_header_;
   std::vector<uint16_t> last_nack_list_;
+  uint8_t keepalive_payload_type_;
+  size_t num_keepalive_sent_;
 };
 
 class RtpRtcpModule : public RtcpPacketTypeCounterObserver {
  public:
   explicit RtpRtcpModule(SimulatedClock* clock)
       : receive_statistics_(ReceiveStatistics::Create(clock)),
         remote_ssrc_(0),
-        retransmission_rate_limiter_(clock, kMaxRttMs) {
-    RtpRtcp::Configuration config;
-    config.audio = false;
-    config.clock = clock;
-    config.outgoing_transport = &transport_;
-    config.receive_statistics = receive_statistics_.get();
-    config.rtcp_packet_type_counter_observer = this;
-    config.rtt_stats = &rtt_stats_;
-    config.retransmission_rate_limiter = &retransmission_rate_limiter_;
-
-    impl_.reset(new ModuleRtpRtcpImpl(config));
-    impl_->SetRTCPStatus(RtcpMode::kCompound);
-
+        retransmission_rate_limiter_(clock, kMaxRttMs),
+        clock_(clock) {
+    CreateModuleImpl();
     transport_.SimulateNetworkDelay(kOneWayNetworkDelayMs, clock);
   }
 
   RtcpPacketTypeCounter packets_sent_;
   RtcpPacketTypeCounter packets_received_;
   std::unique_ptr<ReceiveStatistics> receive_statistics_;
   SendTransport transport_;
   RtcpRttStatsTestImpl rtt_stats_;
   std::unique_ptr<ModuleRtpRtcpImpl> impl_;
   uint32_t remote_ssrc_;
   RateLimiter retransmission_rate_limiter_;
+  VideoSendStream::Config::Rtp::KeepAlive keepalive_config_;
 
   void SetRemoteSsrc(uint32_t ssrc) {
     remote_ssrc_ = ssrc;
     impl_->SetRemoteSSRC(ssrc);
   }
 
   void RtcpPacketTypesCounterUpdated(
       uint32_t ssrc,
       const RtcpPacketTypeCounter& packet_counter) override {
     counter_map_[ssrc] = packet_counter;
@@ skipping 10 matching lines @@
   }
   int RtpSent() {
     return transport_.rtp_packets_sent_;
   }
   uint16_t LastRtpSequenceNumber() {
     return transport_.last_rtp_header_.sequenceNumber;
   }
   std::vector<uint16_t> LastNackListSent() {
     return transport_.last_nack_list_;
   }
+  void SetKeepaliveConfigAndReset(
+      const VideoSendStream::Config::Rtp::KeepAlive& config) {
+    keepalive_config_ = config;
+    // Need to create a new module impl, since it's configured at creation.
+    CreateModuleImpl();
+  }
 
  private:
+  void CreateModuleImpl() {
+    RtpRtcp::Configuration config;
+    config.audio = false;
+    config.clock = clock_;
+    config.outgoing_transport = &transport_;
+    config.receive_statistics = receive_statistics_.get();
+    config.rtcp_packet_type_counter_observer = this;
+    config.rtt_stats = &rtt_stats_;
+    config.retransmission_rate_limiter = &retransmission_rate_limiter_;
+    config.keepalive_config = keepalive_config_;
+
+    impl_.reset(new ModuleRtpRtcpImpl(config));
+    impl_->SetRTCPStatus(RtcpMode::kCompound);
+  }
+
+  SimulatedClock* const clock_;
   std::map<uint32_t, RtcpPacketTypeCounter> counter_map_;
 };
 }  // namespace
 
 class RtpRtcpImplTest : public ::testing::Test {
  protected:
   RtpRtcpImplTest()
-      : clock_(133590000000000),
-        sender_(&clock_),
-        receiver_(&clock_) {
+      : clock_(133590000000000), sender_(&clock_), receiver_(&clock_) {}
+
+  void SetUp() override {
     // Send module.
     sender_.impl_->SetSSRC(kSenderSsrc);
     EXPECT_EQ(0, sender_.impl_->SetSendingStatus(true));
     sender_.impl_->SetSendingMediaStatus(true);
     sender_.SetRemoteSsrc(kReceiverSsrc);
     sender_.impl_->SetSequenceNumber(kSequenceNumber);
     sender_.impl_->SetStorePacketsStatus(true, 100);
 
     memset(&codec_, 0, sizeof(VideoCodec));
     codec_.plType = 100;
     strncpy(codec_.plName, "VP8", 3);
     codec_.width = 320;
     codec_.height = 180;
     EXPECT_EQ(0, sender_.impl_->RegisterSendPayload(codec_));
 
     // Receive module.
     EXPECT_EQ(0, receiver_.impl_->SetSendingStatus(false));
     receiver_.impl_->SetSendingMediaStatus(false);
     receiver_.impl_->SetSSRC(kReceiverSsrc);
     receiver_.SetRemoteSsrc(kSenderSsrc);
     // Transport settings.
     sender_.transport_.SetRtpRtcpModule(receiver_.impl_.get());
     receiver_.transport_.SetRtpRtcpModule(sender_.impl_.get());
   }
+
   SimulatedClock clock_;
   RtpRtcpModule sender_;
   RtpRtcpModule receiver_;
   VideoCodec codec_;
 
   void SendFrame(const RtpRtcpModule* module, uint8_t tid) {
     RTPVideoHeaderVP8 vp8_header = {};
     vp8_header.temporalIdx = tid;
     RTPVideoHeader rtp_video_header;
     rtp_video_header.width = codec_.width;
@@ skipping 351 matching lines @@
   EXPECT_EQ(8U, receiver_.RtcpSent().nack_requests);
   EXPECT_EQ(6U, receiver_.RtcpSent().unique_nack_requests);
   EXPECT_THAT(receiver_.LastNackListSent(), ElementsAre(11, 18, 20, 21));
 
   // Send module receives the request.
   EXPECT_EQ(2U, sender_.RtcpReceived().nack_packets);
   EXPECT_EQ(8U, sender_.RtcpReceived().nack_requests);
   EXPECT_EQ(6U, sender_.RtcpReceived().unique_nack_requests);
   EXPECT_EQ(75, sender_.RtcpReceived().UniqueNackRequestsInPercent());
 }
+
+TEST_F(RtpRtcpImplTest, SendsKeepaliveAfterTimout) {
+  const int kTimeout = 1500;

[åsapersson, 2017/06/30 15:09:13]

kTimeoutMs

[sprang_webrtc, 2017/06/30 16:20:59]

Done.

+
+  VideoSendStream::Config::Rtp::KeepAlive config;
+  config.timeout_interval_ms = kTimeout;
+
+  // Recreate sender impl with new configuration, and redo setup.
+  sender_.SetKeepaliveConfigAndReset(config);
+  SetUp();
+
+  // Initial process call.
+  sender_.impl_->Process();
+  EXPECT_EQ(0U, sender_.transport_.NumKeepaliveSent());
+
+  // After one time, a single keep-alive packet should be sent.
+  clock_.AdvanceTimeMilliseconds(kTimeout);
+  sender_.impl_->Process();
+  EXPECT_EQ(1U, sender_.transport_.NumKeepaliveSent());
+
+  // Process for the same timestamp again, no new packet shold be sent.

[åsapersson, 2017/06/30 15:09:13]

should

[sprang_webrtc, 2017/06/30 16:21:00]

Done.

+  sender_.impl_->Process();
+  EXPECT_EQ(1U, sender_.transport_.NumKeepaliveSent());
+
+  // Move ahead to the last ms before a keep-alive is expected, no action.
+  clock_.AdvanceTimeMilliseconds(kTimeout - 1);
+  sender_.impl_->Process();
+  EXPECT_EQ(1U, sender_.transport_.NumKeepaliveSent());
+
+  // Move the final ms, timeout relative last KA. Should create new keep-alive.
+  clock_.AdvanceTimeMilliseconds(1);
+  sender_.impl_->Process();
+  EXPECT_EQ(2U, sender_.transport_.NumKeepaliveSent());
+
+  // Move ahead to the last ms before Christmas.
+  clock_.AdvanceTimeMilliseconds(kTimeout - 1);
+  sender_.impl_->Process();
+  EXPECT_EQ(2U, sender_.transport_.NumKeepaliveSent());
+
+  // Send actual payload data, no keep-alive expected.
+  SendFrame(&sender_, 0);
+  sender_.impl_->Process();
+  EXPECT_EQ(2U, sender_.transport_.NumKeepaliveSent());
+
+  // Move ahead as far as possible again, timeout now relative payload. No KA.
+  clock_.AdvanceTimeMilliseconds(kTimeout - 1);
+  sender_.impl_->Process();
+  EXPECT_EQ(2U, sender_.transport_.NumKeepaliveSent());
+
+  // Timeout relative payload, send new keep-alive.
+  clock_.AdvanceTimeMilliseconds(1);
+  sender_.impl_->Process();
+  EXPECT_EQ(3U, sender_.transport_.NumKeepaliveSent());
+}
 }  // namespace webrtc