modules_unittests: Turned on ApmTest.Process test for Android

The reason why ApmTest.Process breaks on Android is that two metrics over counts. I decided to add an offset and a different slack to the EXPECT_NEAR() calls that are affected. I think this is a reasonable approach since we have no more than two failing metrics. If any feature change that will make another metric fail, we should go back to the desk and find another way of solving this. BUG=114 TESTED=locally on Nexus 7 and trybots R=aluebs@webrtc.org, andrew@webrtc.org, kwiberg@webrtc.org Review URL: https://webrtc-codereview.appspot.com/26509004 git-svn-id: http://webrtc.googlecode.com/svn/trunk@7268 4adac7df-926f-26a2-2b94-8c16560cd09d
GetStream · Sep 23, 2014 · dc0b37d · dc0b37d
1 parent a3c4d4d
commit dc0b37d
Showing 1 changed file with 29 additions and 10 deletions.
diff --git a/webrtc/modules/audio_processing/test/audio_processing_unittest.cc b/webrtc/modules/audio_processing/test/audio_processing_unittest.cc
@@ -214,6 +214,7 @@ void WriteStatsMessage(const AudioProcessing::Statistic& output,
 
 void OpenFileAndWriteMessage(const std::string filename,
                              const ::google::protobuf::MessageLite& msg) {
+#if defined(WEBRTC_LINUX) && !defined(WEBRTC_ANDROID)
   FILE* file = fopen(filename.c_str(), "wb");
   ASSERT_TRUE(file != NULL);
 
@@ -226,6 +227,10 @@ void OpenFileAndWriteMessage(const std::string filename,
   ASSERT_EQ(static_cast<size_t>(size),
       fwrite(array.get(), sizeof(array[0]), size, file));
   fclose(file);
+#else
+  std::cout << "Warning: Writing new reference is only allowed on Linux!"
+      << std::endl;
+#endif
 }
 
 std::string ResourceFilePath(std::string name, int sample_rate_hz) {
@@ -1735,14 +1740,7 @@ TEST_F(ApmTest, FloatAndIntInterfacesGiveIdenticalResults) {
 // TODO(andrew): Add a test to process a few frames with different combinations
 // of enabled components.
 
-// TODO(bjornv): Investigate if simply increasing the slack is a good way to
-// make this test work on Android. When running the test on a N7 we get a {2, 6}
-// difference of |has_voice_count| and |max_output_average| is up to 18 higher.
-// All numbers being consistently higher on N7 compare to ref_data, evaluated on
-// linux. Simply increasing the slack is one way forward. Adding an offset to
-// the metrics mentioned above, but keeping the same slack, is also an
-// alternative.
-TEST_F(ApmTest, DISABLED_ON_ANDROID(Process)) {
+TEST_F(ApmTest, Process) {
   GOOGLE_PROTOBUF_VERIFY_VERSION;
   audioproc::OutputData ref_data;
 
@@ -1857,12 +1855,33 @@ TEST_F(ApmTest, DISABLED_ON_ANDROID(Process)) {
 
     if (!write_ref_data) {
       const int kIntNear = 1;
+      // When running the test on a N7 we get a {2, 6} difference of
+      // |has_voice_count| and |max_output_average| is up to 18 higher.
+      // All numbers being consistently higher on N7 compare to ref_data.
+      // TODO(bjornv): If we start getting more of these offsets on Android we
+      // should consider a different approach. Either using one slack for all,
+      // or generate a separate android reference.
+#if defined(WEBRTC_ANDROID)
+      const int kHasVoiceCountOffset = 3;
+      const int kHasVoiceCountNear = 3;
+      const int kMaxOutputAverageOffset = 9;
+      const int kMaxOutputAverageNear = 9;
+#else
+      const int kHasVoiceCountOffset = 0;
+      const int kHasVoiceCountNear = kIntNear;
+      const int kMaxOutputAverageOffset = 0;
+      const int kMaxOutputAverageNear = kIntNear;
+#endif
       EXPECT_NEAR(test->has_echo_count(), has_echo_count, kIntNear);
-      EXPECT_NEAR(test->has_voice_count(), has_voice_count, kIntNear);
+      EXPECT_NEAR(test->has_voice_count(),
+                  has_voice_count - kHasVoiceCountOffset,
+                  kHasVoiceCountNear);
       EXPECT_NEAR(test->is_saturated_count(), is_saturated_count, kIntNear);
 
       EXPECT_NEAR(test->analog_level_average(), analog_level_average, kIntNear);
-      EXPECT_NEAR(test->max_output_average(), max_output_average, kIntNear);
+      EXPECT_NEAR(test->max_output_average(),
+                  max_output_average - kMaxOutputAverageOffset,
+                  kMaxOutputAverageNear);
 
 #if defined(WEBRTC_AUDIOPROC_FLOAT_PROFILE)
       audioproc::Test::EchoMetrics reference = test->echo_metrics();