ViE autotest: Adding loss models to the external transport

Review URL: https://webrtc-codereview.appspot.com/1000004

git-svn-id: http://webrtc.googlecode.com/svn/trunk@3309 4adac7df-926f-26a2-2b94-8c16560cd09d

webrtc/video_engine/test/auto_test/automated/vie_video_verification_test.cc

@@ -125,7 +125,10 @@ class ParameterizedFullStackTest : public ViEVideoVerificationTest,
 
   void SetUp() {
     int i = 0;
+    // Uniform loss => Setting burst length to -1.
+    parameter_table_[i].network.loss_model = kUniformLoss;
     parameter_table_[i].network.packet_loss_rate = 0;
+    parameter_table_[i].network.burst_length = -1;
     parameter_table_[i].network.mean_one_way_delay = 0;
     parameter_table_[i].network.std_dev_one_way_delay = 0;
     parameter_table_[i].bitrate = 300;
@@ -133,7 +136,9 @@ class ParameterizedFullStackTest : public ViEVideoVerificationTest,
     parameter_table_[i].avg_ssim_threshold = 0.96;
     parameter_table_[i].test_label = "net delay (0, 0), plr 0";
     ++i;
+    parameter_table_[i].network.loss_model = kUniformLoss;
     parameter_table_[i].network.packet_loss_rate = 5;
+    parameter_table_[i].network.burst_length = -1;
     parameter_table_[i].network.mean_one_way_delay = 50;
     parameter_table_[i].network.std_dev_one_way_delay = 5;
     parameter_table_[i].bitrate = 300;
@@ -141,7 +146,19 @@ class ParameterizedFullStackTest : public ViEVideoVerificationTest,
     parameter_table_[i].avg_ssim_threshold = 0.96;
     parameter_table_[i].test_label = "net delay (50, 5), plr 5";
     ++i;
+    parameter_table_[i].network.loss_model = kUniformLoss;
     parameter_table_[i].network.packet_loss_rate = 0;
+    parameter_table_[i].network.burst_length = -1;
+    parameter_table_[i].network.mean_one_way_delay = 100;
+    parameter_table_[i].network.std_dev_one_way_delay = 10;
+    parameter_table_[i].bitrate = 300;
+    parameter_table_[i].avg_psnr_threshold = 35;
+    parameter_table_[i].avg_ssim_threshold = 0.96;
+    parameter_table_[i].test_label = "net delay (100, 10), plr 0";
+    ++i;
+    parameter_table_[i].network.loss_model = kGilbertElliotLoss;
+    parameter_table_[i].network.packet_loss_rate = 5;
+    parameter_table_[i].network.burst_length = 3;
     parameter_table_[i].network.mean_one_way_delay = 100;
     parameter_table_[i].network.std_dev_one_way_delay = 10;
     parameter_table_[i].bitrate = 300;
@@ -150,7 +167,7 @@ class ParameterizedFullStackTest : public ViEVideoVerificationTest,
     parameter_table_[i].test_label = "net delay (100, 10), plr 0";
   }
 
-  TestParameters parameter_table_[3];
+  TestParameters parameter_table_[4];
 };
 
 TEST_F(ViEVideoVerificationTest, RunsBaseStandardTestWithoutErrors) {
@@ -207,6 +224,7 @@ TEST_P(ParameterizedFullStackTest, RunsFullStackWithoutErrors)  {
   // frames every now and then.
   const int kBitRateKbps = parameter_table_[GetParam()].bitrate;
   const NetworkParameters network = parameter_table_[GetParam()].network;
+  // TODO(mikhal): Resolution and file name should be in the parameter list.
   int width = 352;
   int height = 288;
   ViETest::Log("Bit rate     : %5d kbps", kBitRateKbps);

webrtc/video_engine/test/libvietest/include/tb_external_transport.h

@@ -28,10 +28,17 @@ class ThreadWrapper;
 class ViENetwork;
 }
 
+enum RandomLossModel {
+  kNoLoss,
+  kUniformLoss,
+  kGilbertElliotLoss
+};
 struct NetworkParameters {
   int packet_loss_rate;
+  int burst_length;  // Only applicable for kGilbertElliotLoss.
   int mean_one_way_delay;
   int std_dev_one_way_delay;
+  RandomLossModel loss_model;
 };
 
 // Allows to subscribe for callback when a frame is started being sent.
@@ -109,7 +116,10 @@ protected:
     static bool ViEExternalTransportRun(void* object);
     bool ViEExternalTransportProcess();
 private:
+    // TODO(mikhal): Break these out to classes.
     static int GaussianRandom(int mean_ms, int standard_deviation_ms);
+    bool UniformLoss(int loss_rate);
+    bool GilbertElliotLoss(int loss_rate, int burst_length);
     WebRtc_Word64 NowMs();
 
     enum
@@ -169,6 +179,7 @@ private:
     WebRtc_UWord32 _lastSendRTPTimestamp;
     WebRtc_UWord32 _lastReceiveRTPTimestamp;
     int64_t last_receive_time_;
+    bool previous_drop_;
 };
 
 #endif  // WEBRTC_VIDEO_ENGINE_TEST_AUTOTEST_INTERFACE_TB_EXTERNAL_TRANSPORT_H_

webrtc/video_engine/test/libvietest/testbed/tb_external_transport.cc

@@ -73,7 +73,8 @@ TbExternalTransport::TbExternalTransport(
       _firstRTPTimestamp(0),
       _lastSendRTPTimestamp(0),
       _lastReceiveRTPTimestamp(0),
-      last_receive_time_(-1)
+      last_receive_time_(-1),
+      previous_drop_(false)
 {
     srand((int) webrtc::TickTime::MicrosecondTimestamp());
     unsigned int tId = 0;
@@ -190,11 +191,24 @@ int TbExternalTransport::SendPacket(int channel, const void *data, int len)
     _rtpCount++;
     _statCrit.Leave();
 
-    // Packet loss. Never drop packets from the first RTP timestamp, i.e. the
+    // Packet loss.
-    // first frame being transmitted.
+    switch (network_parameters_.loss_model)
-    int dropThis = rand() % 100;
+    {
-    if (dropThis < network_parameters_.packet_loss_rate &&
+        case (kNoLoss):
-        _firstRTPTimestamp != rtp_timestamp)
+            previous_drop_ = false;
+            break;
+        case (kUniformLoss):
+            previous_drop_ = UniformLoss(network_parameters_.packet_loss_rate);
+            break;
+        case (kGilbertElliotLoss):
+            previous_drop_ = GilbertElliotLoss(
+                network_parameters_.packet_loss_rate,
+                network_parameters_.burst_length);
+            break;
+    }
+    // Never drop packets from the first RTP timestamp (first frame)
+    // transmitted.
+    if (previous_drop_ && _firstRTPTimestamp != rtp_timestamp)
     {
         _statCrit.Enter();
         _dropCount++;
@@ -501,6 +515,42 @@ WebRtc_Word64 TbExternalTransport::NowMs()
     return webrtc::TickTime::MillisecondTimestamp();
 }
 
+bool TbExternalTransport::UniformLoss(int loss_rate) {
+  int dropThis = rand() % 100;
+  return (dropThis < loss_rate);
+}
+
+bool TbExternalTransport::GilbertElliotLoss(int loss_rate, int burst_length) {
+  // Simulate bursty channel (Gilbert model)
+  // (1st order) Markov chain model with memory of the previous/last
+  // packet state (loss or received)
+
+  // 0 = received state
+  // 1 = loss state
+
+  // probTrans10: if previous packet is lost, prob. to -> received state
+  // probTrans11: if previous packet is lost, prob. to -> loss state
+
+  // probTrans01: if previous packet is received, prob. to -> loss state
+  // probTrans00: if previous packet is received, prob. to -> received
+
+  // Map the two channel parameters (average loss rate and burst length)
+  // to the transition probabilities:
+  double probTrans10 = 100 * (1.0 / burst_length);
+  double probTrans11 = (100.0 - probTrans10);
+  double probTrans01 = (probTrans10 * ( loss_rate / (100.0 - loss_rate)));
+
+  // Note: Random loss (Bernoulli) model is a special case where:
+  // burstLength = 100.0 / (100.0 - _lossPct) (i.e., p10 + p01 = 100)
+
+  if (previous_drop_) {
+    // Previous packet was not received.
+    return UniformLoss(probTrans11);
+  } else {
+    return UniformLoss(probTrans01);
+  }
+}
+
 #define PI  3.14159265
 int TbExternalTransport::GaussianRandom(int mean_ms,
                                         int standard_deviation_ms) {