Add a BWE fairness test.

Also moves the BWE perf tests to webrtc_perf_tests for tracking.

BUG=4548
R=pbos@webrtc.org

Review URL: https://webrtc-codereview.appspot.com/45189004

Cr-Commit-Position: refs/heads/master@{#9022}

webrtc/modules/modules.gyp

@@ -60,6 +60,7 @@
             'audio_device'  ,
             'audio_processing',
             'bitrate_controller',
+            'bwe_simulator',
             'CNG',
             'desktop_capture',
             'iSACFix',
@@ -216,33 +217,6 @@
             'remote_bitrate_estimator/remote_bitrate_estimator_single_stream_unittest.cc',
             'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.cc',
             'remote_bitrate_estimator/remote_bitrate_estimator_unittest_helper.h',
-            'remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
-            'remote_bitrate_estimator/test/bwe.cc',
-            'remote_bitrate_estimator/test/bwe.h',
-            'remote_bitrate_estimator/test/bwe_test.cc',
-            'remote_bitrate_estimator/test/bwe_test.h',
-            'remote_bitrate_estimator/test/bwe_test_baselinefile.cc',
-            'remote_bitrate_estimator/test/bwe_test_baselinefile.h',
-            'remote_bitrate_estimator/test/bwe_test_fileutils.cc',
-            'remote_bitrate_estimator/test/bwe_test_fileutils.h',
-            'remote_bitrate_estimator/test/bwe_test_framework.cc',
-            'remote_bitrate_estimator/test/bwe_test_framework.h',
-            'remote_bitrate_estimator/test/bwe_test_framework_unittest.cc',
-            'remote_bitrate_estimator/test/bwe_test_logging.cc',
-            'remote_bitrate_estimator/test/bwe_test_logging.h',
-            'remote_bitrate_estimator/test/packet_receiver.cc',
-            'remote_bitrate_estimator/test/packet_receiver.h',
-            'remote_bitrate_estimator/test/packet_sender.cc',
-            'remote_bitrate_estimator/test/packet_sender.h',
-            'remote_bitrate_estimator/test/packet.h',
-            'remote_bitrate_estimator/test/estimators/nada.cc',
-            'remote_bitrate_estimator/test/estimators/nada.h',
-            'remote_bitrate_estimator/test/estimators/remb.cc',
-            'remote_bitrate_estimator/test/estimators/remb.h',
-            'remote_bitrate_estimator/test/estimators/send_side.cc',
-            'remote_bitrate_estimator/test/estimators/send_side.h',
-            'remote_bitrate_estimator/test/estimators/tcp.cc',
-            'remote_bitrate_estimator/test/estimators/tcp.h',
             'rtp_rtcp/source/mock/mock_rtp_payload_strategy.h',
             'rtp_rtcp/source/byte_io_unittest.cc',
             'rtp_rtcp/source/fec_receiver_unittest.cc',

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimator.gypi

@@ -55,6 +55,41 @@
   'conditions': [
     ['include_tests==1', {
       'targets': [
+        {
+          'target_name': 'bwe_simulator',
+          'type': 'static_library',
+          'dependencies': [
+            '<(DEPTH)/testing/gtest.gyp:gtest',
+          ],
+          'sources': [
+            'test/bwe.cc',
+            'test/bwe.h',
+            'test/bwe_test.cc',
+            'test/bwe_test.h',
+            'test/bwe_test_baselinefile.cc',
+            'test/bwe_test_baselinefile.h',
+            'test/bwe_test_fileutils.cc',
+            'test/bwe_test_fileutils.h',
+            'test/bwe_test_framework.cc',
+            'test/bwe_test_framework.h',
+            'test/bwe_test_framework_unittest.cc',
+            'test/bwe_test_logging.cc',
+            'test/bwe_test_logging.h',
+            'test/packet_receiver.cc',
+            'test/packet_receiver.h',
+            'test/packet_sender.cc',
+            'test/packet_sender.h',
+            'test/packet.h',
+            'test/estimators/nada.cc',
+            'test/estimators/nada.h',
+            'test/estimators/remb.cc',
+            'test/estimators/remb.h',
+            'test/estimators/send_side.cc',
+            'test/estimators/send_side.h',
+            'test/estimators/tcp.cc',
+            'test/estimators/tcp.h',
+          ],
+        },
         {
           'target_name': 'bwe_tools_util',
           'type': 'static_library',

webrtc/modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc

@@ -241,8 +241,10 @@ class BweFeedbackTest
   BweFeedbackTest() : BweTest() {}
   virtual ~BweFeedbackTest() {}
 
-  void PrintResults(double max_throughput_kbps, Stats<double> throughput_kbps,
+  void PrintResults(double max_throughput_kbps,
-                    Stats<double> delay_ms) {
+                    Stats<double> throughput_kbps,
+                    Stats<double> delay_ms,
+                    std::vector<Stats<double>> flow_throughput_kbps) {
     double utilization = throughput_kbps.GetMean() / max_throughput_kbps;
     webrtc::test::PrintResult("BwePerformance",
                               GetTestName(),
@@ -264,6 +266,16 @@ class BweFeedbackTest
                               delay_ms.AsString(),
                               "ms",
                               false);
+    double fairness_index = 0.0;
+    double squared_bitrate_sum = 0.0;
+    for (Stats<double> flow : flow_throughput_kbps) {
+      squared_bitrate_sum += flow.GetMean() * flow.GetMean();
+      fairness_index += flow.GetMean();
+    }
+    fairness_index *= fairness_index;
+    fairness_index /= flow_throughput_kbps.size() * squared_bitrate_sum;
+    webrtc::test::PrintResult("BwePerformance", GetTestName(), "Fairness",
+                              fairness_index * 100, "%", false);
   }
 
  protected:
@@ -294,7 +306,8 @@ TEST_P(BweFeedbackTest, Choke1000kbps500kbps1000kbps) {
   filter.SetCapacity(kHighCapacityKbps);
   RunFor(60 * 1000);
   PrintResults((2 * kHighCapacityKbps + kLowCapacityKbps) / 3.0,
-               counter.GetBitrateStats(), filter.GetDelayStats());
+               counter.GetBitrateStats(), filter.GetDelayStats(),
+               std::vector<Stats<double>>());
 }
 
 TEST_P(BweFeedbackTest, Choke200kbps30kbps200kbps) {
@@ -314,7 +327,8 @@ TEST_P(BweFeedbackTest, Choke200kbps30kbps200kbps) {
   RunFor(60 * 1000);
 
   PrintResults((2 * kHighCapacityKbps + kLowCapacityKbps) / 3.0,
-               counter.GetBitrateStats(), filter.GetDelayStats());
+               counter.GetBitrateStats(), filter.GetDelayStats(),
+               std::vector<Stats<double>>());
 }
 
 TEST_P(BweFeedbackTest, Verizon4gDownlinkTest) {
@@ -327,7 +341,7 @@ TEST_P(BweFeedbackTest, Verizon4gDownlinkTest) {
   ASSERT_TRUE(filter.Init(test::ResourcePath("verizon4g-downlink", "rx")));
   RunFor(22 * 60 * 1000);
   PrintResults(filter.GetBitrateStats().GetMean(), counter2.GetBitrateStats(),
-               filter.GetDelayStats());
+               filter.GetDelayStats(), std::vector<Stats<double>>());
 }
 
 // webrtc:3277
@@ -342,7 +356,50 @@ TEST_P(BweFeedbackTest, DISABLED_GoogleWifiTrace3Mbps) {
   ASSERT_TRUE(filter.Init(test::ResourcePath("google-wifi-3mbps", "rx")));
   RunFor(300 * 1000);
   PrintResults(filter.GetBitrateStats().GetMean(), counter2.GetBitrateStats(),
-               filter.GetDelayStats());
+               filter.GetDelayStats(), std::vector<Stats<double>>());
+}
+
+TEST_P(BweFeedbackTest, PacedSelfFairnessTest) {
+  srand(Clock::GetRealTimeClock()->TimeInMicroseconds());
+  const int kAllFlowIds[] = {0, 1, 2, 3};
+  const size_t kNumFlows = sizeof(kAllFlowIds) / sizeof(kAllFlowIds[0]);
+  rtc::scoped_ptr<VideoSource> sources[kNumFlows];
+  rtc::scoped_ptr<PacketSender> senders[kNumFlows];
+
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    // Streams started woth ramdp, pffsets tp give them different advantage when
+    // competing for the bandwidth.
+    sources[i].reset(new AdaptiveVideoSource(kAllFlowIds[i], 30, 300, 0,
+                                             i * (rand() % 40000)));
+    senders[i].reset(
+        new PacedVideoSender(&uplink_, sources[i].get(), GetParam()));
+  }
+
+  ChokeFilter choke(&uplink_, CreateFlowIds(kAllFlowIds, kNumFlows));
+  choke.SetCapacity(3000);
+  choke.SetMaxDelay(1000);
+
+  rtc::scoped_ptr<RateCounterFilter> rate_counters[kNumFlows];
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    rate_counters[i].reset(new RateCounterFilter(
+        &uplink_, CreateFlowIds(&kAllFlowIds[i], 1), "receiver_input"));
+  }
+
+  RateCounterFilter total_utilization(
+      &uplink_, CreateFlowIds(kAllFlowIds, kNumFlows), "total_utilization");
+
+  rtc::scoped_ptr<PacketReceiver> receivers[kNumFlows];
+  for (size_t i = 0; i < kNumFlows; ++i) {
+    receivers[i].reset(new PacketReceiver(&uplink_, kAllFlowIds[i], GetParam(),
+                                          i == 0, false));
+  }
+  RunFor(15 * 60 * 1000);
+
+  std::vector<Stats<double>> flow_throughput_kbps;
+  for (size_t i = 0; i < kNumFlows; ++i)
+    flow_throughput_kbps.push_back(rate_counters[i]->GetBitrateStats());
+  PrintResults(3000, total_utilization.GetBitrateStats(), choke.GetDelayStats(),
+               flow_throughput_kbps);
 }
 }  // namespace bwe
 }  // namespace testing

webrtc/modules/remote_bitrate_estimator/test/bwe_test_framework_unittest.cc

@@ -615,7 +615,7 @@ class BweTestFramework_ChokeFilterTest : public ::testing::Test {
   void TestChoke(PacketProcessor* filter,
                  int64_t run_for_ms,
                  uint32_t packets_to_generate,
-                 uint32_t expected_kbit_transmitted) {
+                 size_t expected_kbit_transmitted) {
     // Generate a bunch of packets, apply choke, verify output is ordered.
     Packets packets;
     RTPHeader header;
@@ -634,7 +634,7 @@ class BweTestFramework_ChokeFilterTest : public ::testing::Test {
     ASSERT_TRUE(IsSequenceNumberSorted(output_packets_));
 
     // Sum up the transmitted bytes up until the current time.
-    uint32_t bytes_transmitted = 0;
+    size_t bytes_transmitted = 0;
     while (!output_packets_.empty()) {
       const Packet* packet = output_packets_.front();
       if (packet->send_time_us() > now_ms_ * 1000) {
@@ -769,7 +769,7 @@ void TestVideoSender(VideoSender* sender,
                      int64_t run_for_ms,
                      uint32_t expected_packets,
                      uint32_t expected_payload_size,
-                     uint32_t expected_total_payload_size) {
+                     size_t expected_total_payload_size) {
   assert(sender);
   Packets packets;
   sender->RunFor(run_for_ms, &packets);
@@ -777,7 +777,7 @@ void TestVideoSender(VideoSender* sender,
   ASSERT_TRUE(IsSequenceNumberSorted(packets));
   EXPECT_EQ(expected_packets, packets.size());
   int64_t send_time_us = -1;
-  uint32_t total_payload_size = 0;
+  size_t total_payload_size = 0;
   uint32_t absolute_send_time = 0;
   uint32_t absolute_send_time_wraps = 0;
   uint32_t rtp_timestamp = 0;

webrtc/webrtc_tests.gypi

@@ -178,6 +178,8 @@
       'type': '<(gtest_target_type)',
       'sources': [
         'modules/audio_coding/neteq/test/neteq_performance_unittest.cc',
+        'modules/remote_bitrate_estimator/remote_bitrate_estimators_test.cc',
+
         'tools/agc/agc_manager_integrationtest.cc',
         'video/call_perf_tests.cc',
         'video/full_stack.cc',
@@ -190,7 +192,8 @@
         '<(webrtc_root)/modules/modules.gyp:video_capture',
         '<(webrtc_root)/test/test.gyp:channel_transport',
         '<(webrtc_root)/voice_engine/voice_engine.gyp:voice_engine',
-        'modules/modules.gyp:neteq_test_support',  # Needed by neteq_performance_unittest.
+        'modules/modules.gyp:neteq_test_support',
+        'modules/modules.gyp:bwe_simulator',
         'modules/modules.gyp:rtp_rtcp',
         'test/test.gyp:test_main',
         'test/webrtc_test_common.gyp:webrtc_test_common',