diff --git a/webrtc/call/call_perf_tests.cc b/webrtc/call/call_perf_tests.cc
index 56c6ba3535..ae092f3837 100644
--- a/webrtc/call/call_perf_tests.cc
+++ b/webrtc/call/call_perf_tests.cc
@@ -288,7 +288,11 @@ void CallPerfTest::TestAudioVideoSync(FecMode fec,
   VoiceEngine::Delete(voice_engine);
 
   observer.PrintResults();
-  EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
+
+  // In quick test synchronization may not be achieved in time.
+  if (field_trial::FindFullName("WebRTC-QuickPerfTest") != "Enabled") {
+    EXPECT_EQ(1, metrics::NumSamples("WebRTC.Video.AVSyncOffsetInMs"));
+  }
 }
 
 TEST_F(CallPerfTest, PlaysOutAudioAndVideoInSyncWithVideoNtpDrift) {
diff --git a/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc b/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
index cd5ce4e576..23f43b5426 100644
--- a/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
+++ b/webrtc/modules/audio_coding/neteq/test/neteq_performance_unittest.cc
@@ -12,15 +12,20 @@
 #include "webrtc/test/gtest.h"
 #include "webrtc/test/testsupport/perf_test.h"
 #include "webrtc/typedefs.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 
 // Runs a test with 10% packet losses and 10% clock drift, to exercise
 // both loss concealment and time-stretching code.
 TEST(NetEqPerformanceTest, Run) {
   const int kSimulationTimeMs = 10000000;
+  const int kQuickSimulationTimeMs = 100000;
   const int kLossPeriod = 10;  // Drop every 10th packet.
   const double kDriftFactor = 0.1;
   int64_t runtime = webrtc::test::NetEqPerformanceTest::Run(
-      kSimulationTimeMs, kLossPeriod, kDriftFactor);
+      webrtc::field_trial::FindFullName("WebRTC-QuickPerfTest") == "Enabled"
+          ? kQuickSimulationTimeMs
+          : kSimulationTimeMs,
+      kLossPeriod, kDriftFactor);
   ASSERT_GT(runtime, 0);
   webrtc::test::PrintResult(
       "neteq_performance", "", "10_pl_10_drift", runtime, "ms", true);
@@ -31,10 +36,14 @@ TEST(NetEqPerformanceTest, Run) {
 // more lightweight.
 TEST(NetEqPerformanceTest, RunClean) {
   const int kSimulationTimeMs = 10000000;
+  const int kQuickSimulationTimeMs = 100000;
   const int kLossPeriod = 0;  // No losses.
   const double kDriftFactor = 0.0;  // No clock drift.
   int64_t runtime = webrtc::test::NetEqPerformanceTest::Run(
-      kSimulationTimeMs, kLossPeriod, kDriftFactor);
+      webrtc::field_trial::FindFullName("WebRTC-QuickPerfTest") == "Enabled"
+          ? kQuickSimulationTimeMs
+          : kSimulationTimeMs,
+      kLossPeriod, kDriftFactor);
   ASSERT_GT(runtime, 0);
   webrtc::test::PrintResult(
       "neteq_performance", "", "0_pl_0_drift", runtime, "ms", true);
diff --git a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
index ca6563bdca..b243cacf0f 100644
--- a/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
+++ b/webrtc/modules/remote_bitrate_estimator/test/bwe_test.cc
@@ -20,10 +20,15 @@
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_receiver.h"
 #include "webrtc/modules/remote_bitrate_estimator/test/packet_sender.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/test/testsupport/perf_test.h"
 
 using std::vector;
 
+namespace {
+const int kQuickTestTimeoutMs = 500;
+}
+
 namespace webrtc {
 namespace testing {
 namespace bwe {
@@ -160,6 +165,11 @@ void BweTest::VerboseLogging(bool enable) {
 void BweTest::RunFor(int64_t time_ms) {
   // Set simulation interval from first packet sender.
   // TODO(holmer): Support different feedback intervals for different flows.
+
+  // For quick perf tests ignore passed timeout
+  if (field_trial::FindFullName("WebRTC-QuickPerfTest") == "Enabled") {
+    time_ms = kQuickTestTimeoutMs;
+  }
   if (!uplink_.senders().empty()) {
     simulation_interval_ms_ = uplink_.senders()[0]->GetFeedbackIntervalMs();
   } else if (!downlink_.senders().empty()) {
@@ -370,21 +380,23 @@ void BweTest::RunFairnessTest(BandwidthEstimatorType bwe_type,
   PrintResults(capacity_kbps, total_utilization.GetBitrateStats(),
                flow_delay_ms, flow_throughput_kbps);
 
-  for (int i : all_flow_ids) {
-    metric_recorders[i]->PlotThroughputHistogram(
-        title, flow_name, static_cast<int>(num_media_flows), 0);
+  if (field_trial::FindFullName("WebRTC-QuickPerfTest") != "Enabled") {
+    for (int i : all_flow_ids) {
+      metric_recorders[i]->PlotThroughputHistogram(
+          title, flow_name, static_cast<int>(num_media_flows), 0);
 
-    metric_recorders[i]->PlotLossHistogram(title, flow_name,
-                                           static_cast<int>(num_media_flows),
-                                           receivers[i]->GlobalPacketLoss());
-  }
+      metric_recorders[i]->PlotLossHistogram(title, flow_name,
+                                             static_cast<int>(num_media_flows),
+                                             receivers[i]->GlobalPacketLoss());
+    }
 
-  // Pointless to show delay histogram for TCP flow.
-  for (int i : media_flow_ids) {
-    metric_recorders[i]->PlotDelayHistogram(title, bwe_names[bwe_type],
-                                            static_cast<int>(num_media_flows),
-                                            one_way_delay_ms);
-    BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], one_way_delay_ms, i);
+    // Pointless to show delay histogram for TCP flow.
+    for (int i : media_flow_ids) {
+      metric_recorders[i]->PlotDelayHistogram(title, bwe_names[bwe_type],
+                                              static_cast<int>(num_media_flows),
+                                              one_way_delay_ms);
+      BWE_TEST_LOGGING_BASELINEBAR(5, bwe_names[bwe_type], one_way_delay_ms, i);
+    }
   }
 
   for (VideoSource* source : sources)
diff --git a/webrtc/test/rtp_rtcp_observer.h b/webrtc/test/rtp_rtcp_observer.h
index 63e97be4dd..4856de7905 100644
--- a/webrtc/test/rtp_rtcp_observer.h
+++ b/webrtc/test/rtp_rtcp_observer.h
@@ -17,12 +17,17 @@
 #include "webrtc/base/criticalsection.h"
 #include "webrtc/base/event.h"
 #include "webrtc/modules/rtp_rtcp/include/rtp_header_parser.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/test/constants.h"
 #include "webrtc/test/direct_transport.h"
 #include "webrtc/test/gtest.h"
 #include "webrtc/typedefs.h"
 #include "webrtc/video_send_stream.h"
 
+namespace {
+const int kShortTimeoutMs = 500;
+}
+
 namespace webrtc {
 namespace test {
 
@@ -37,7 +42,13 @@ class RtpRtcpObserver {
 
   virtual ~RtpRtcpObserver() {}
 
-  virtual bool Wait() { return observation_complete_.Wait(timeout_ms_); }
+  virtual bool Wait() {
+    if (field_trial::FindFullName("WebRTC-QuickPerfTest") == "Enabled") {
+      observation_complete_.Wait(kShortTimeoutMs);
+      return true;
+    }
+    return observation_complete_.Wait(timeout_ms_);
+  }
 
   virtual Action OnSendRtp(const uint8_t* packet, size_t length) {
     return SEND_PACKET;