added WebRTC-QuickPerfTest to RampUpTests and CallPerfTests

webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc

Index: webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
diff --git a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
index 4349a70390355278c8b38ef34216896967c12bc3..885026698dc966d3ccd2a4484c745569fadb9849 100644
--- a/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
+++ b/webrtc/modules/audio_coding/neteq/tools/neteq_performance_test.cc
@@ -18,6 +18,7 @@
 #include "webrtc/modules/audio_coding/neteq/tools/rtp_generator.h"
 #include "webrtc/modules/include/module_common_types.h"
 #include "webrtc/system_wrappers/include/clock.h"
+#include "webrtc/system_wrappers/include/field_trial.h"
 #include "webrtc/test/testsupport/fileutils.h"
 #include "webrtc/typedefs.h"
 
@@ -26,12 +27,19 @@ using webrtc::test::AudioLoop;
 using webrtc::test::RtpGenerator;
 using webrtc::WebRtcRTPHeader;
 
+namespace {
+  const int kQuickTestTimeoutMs = 100000;

[stefan-webrtc, 2017/02/21 09:35:46]

Sounds like a lot to have 100 seconds for a quick test run?

[ilnik, 2017/02/21 09:46:08]

The variable is named runtime_ms, but it is not a realtime duration, but
simulated runtime. No clocks are tied to it. It turns out, simulation is ~300
times faster than realtime. Originally, tests are run with 10 million ms there.
This value of 100000 is already 100 times less than that.

[stefan-webrtc, 2017/02/21 10:26:03]

Ok

I would prefer if you moved this to where the Run() method is being called, so
you can do:

Run(field_trial::FindFullName("WebRTC-QuickPerfTest") == "Enabled" ?
kQuickTestTimeoutMs : kRegularTestTimeoutMs, loss_rate, drift_factor);

That way the caller won't be surprised.

[hlundin-webrtc, 2017/02/21 10:35:04]

Yes, agree with Stefan. This is an unexpected override of the passed parameter.

+}
+
 namespace webrtc {
 namespace test {
 
 int64_t NetEqPerformanceTest::Run(int runtime_ms,
                                   int lossrate,
                                   double drift_factor) {
+  if (field_trial::FindFullName("WebRTC-QuickPerfTest") == "Enabled") {
+    runtime_ms = kQuickTestTimeoutMs;
+  }
   const std::string kInputFileName =
       webrtc::test::ResourcePath("audio_coding/testfile32kHz", "pcm");
   const int kSampRateHz = 32000;