Deleted the unstable not-silent audio test, moved the quality test to the quality suite.

TBR=dennisjeffrey@chromium.org
NOTRY=True
BUG=


Review URL: https://chromiumcodereview.appspot.com/12566006

git-svn-id: svn://svn.chromium.org/chrome/trunk/src@186716 0039d316-1c4b-4281-b951-d872f2087c98

2 files changed, 10 insertions, 55 deletions

diff --git a/chrome/test/functional/PYAUTO_TESTS b/chrome/test/functional/PYAUTO_TESTS
index 6019125..0db6556 100644
--- a/chrome/test/functional/PYAUTO_TESTS
+++ b/chrome/test/functional/PYAUTO_TESTS
@@ -498,20 +498,13 @@
   },
 
   # WebRTC MediaStream tests. Requires webcam and audio device to be present on
-  # the test machine. The webrtc_audio_call test requires additional
-  # configuration as described in the test's class comments.
+  # the test machine.
   'WEBRTC': {
     'all': [
       'media_stream_infobar',
       'webrtc_brutality_test',
       'webrtc_call',
     ],
-    'linux': [
-      'webrtc_audio_call',
-    ],
-    'win': [
-      'webrtc_audio_call.testWebrtcAudioCallAndMeasureQuality',
-    ],
   },
 
   # WebRTC / AppRTC Integration tests.
@@ -521,11 +514,17 @@
     ],
   },
 
-  # WebRTC quality tests. Requires webcam, an audio device and a special setup
-  # where the webcam is playing a barcode-encoded video.
+  # WebRTC quality tests. The video quality test requires a webcam, an audio
+  # device and a special setup where the webcam plays a barcode-encoded video.
+  # The webrtc_audio_quality test requires additional configuration as described
+  # in the test's class comments.
   'WEBRTC_QUALITY': {
     'all': [
       'webrtc_video_quality',
+      'webrtc_audio_quality',
+    ],
+    'mac': [
+      '-webrtc_audio_quality',  # Not implemented.
     ],
   },
 
diff --git a/chrome/test/functional/webrtc_audio_call.py b/chrome/test/functional/webrtc_audio_quality.py
index a9ba7e9..1064936 100755
--- a/chrome/test/functional/webrtc_audio_call.py
+++ b/chrome/test/functional/webrtc_audio_quality.py
@@ -20,10 +20,9 @@ _MEDIA_PATH = os.path.abspath(os.path.join(pyauto.PyUITest.DataDir(),
 _REFERENCE_FILE = os.path.join(_MEDIA_PATH, 'human-voice.wav')
 _JAVASCRIPT_PATH = os.path.abspath(os.path.join(pyauto.PyUITest.DataDir(),
                                                 'webrtc'))
-_SIZE_OF_EMPTY_WAV_FILE_BYTES = 44
 
 
-class WebrtcAudioCallTest(webrtc_test_base.WebrtcTestBase):
+class WebrtcAudioQualityTest(webrtc_test_base.WebrtcTestBase):
   """Test we can set up a WebRTC call and play audio through it.
 
   This test will only work on machines that have been configured to record their
@@ -64,33 +63,6 @@ class WebrtcAudioCallTest(webrtc_test_base.WebrtcTestBase):
     pyauto.PyUITest.tearDown(self)
     self.assertEquals('', self.CheckErrorsAndCrashes())
 
-  def testWebrtcAudioCallAndVerifyAudioIsPlaying(self):
-    """Test that WebRTC is capable of transmitting at least some audio.
-
-    This test has some nontrivial prerequisites:
-    The target system must have an active microphone, it must be selected
-    as default input for the user that runs the test, and it must record a
-    certain minimum level of ambient noise (for instance server fans).
-    Verify that you are getting ambient noise in the microphone by either
-    recording it directly or checking your OS' microphone settings. Amplify
-    the microphone if the background noise is too low. The microphone should
-    capture noise consistently above 5% of its total range.
-    """
-    self.assertTrue(self.IsLinux(), msg='Only supported on Linux.')
-
-    def CallWithMic():
-      self._AudioCallWithGetUserMedia(duration_seconds=5)
-
-    def EnsureNotAllSilent(output_no_silence):
-      self.assertTrue(os.path.getsize(output_no_silence) >
-                      _SIZE_OF_EMPTY_WAV_FILE_BYTES,
-                      msg=('The test recorded only silence. Ensure your '
-                           'machine is correctly configured for this test.'))
-
-    self._RecordAndVerify(record_duration_seconds=10,
-                          sound_producing_function=CallWithMic,
-                          verification_function=EnsureNotAllSilent)
-
   def testWebrtcAudioCallAndMeasureQuality(self):
     """Measures how much WebRTC distorts speech.
 
@@ -128,22 +100,6 @@ class WebrtcAudioCallTest(webrtc_test_base.WebrtcTestBase):
                           sound_producing_function=CallWithWebAudio,
                           verification_function=MeasureQuality)
 
-  def _AudioCallWithGetUserMedia(self, duration_seconds):
-    self.LoadTestPageInTwoTabs()
-
-    # This sets up a audio-only call.
-    self.assertEquals('ok-got-stream', self.GetUserMedia(tab_index=0,
-                                                         request_video=False))
-    self.assertEquals('ok-got-stream', self.GetUserMedia(tab_index=1,
-                                                         request_video=False))
-    self.Connect('user_1', tab_index=0)
-    self.Connect('user_2', tab_index=1)
-
-    self.CreatePeerConnection(tab_index=0)
-    self.AddUserMediaLocalStream(tab_index=0)
-
-    self._CallAndRunFor(duration_seconds)
-
   def _AudioCallWithWebAudio(self, duration_seconds, input_relative_path):
     self.LoadTestPageInTwoTabs(test_page='webrtc_audio_quality_test.html');