Enable audio capture on Android

content/renderer/media/webrtc_audio_device_unittest.cc

Index: content/renderer/media/webrtc_audio_device_unittest.cc
===================================================================
--- content/renderer/media/webrtc_audio_device_unittest.cc	(revision 181622)
+++ content/renderer/media/webrtc_audio_device_unittest.cc	(working copy)
@@ -322,6 +322,10 @@
 // to send encoded packets to the network. Our main interest here is to ensure
 // that the audio capturing starts as it should.
 // Disabled when running headless since the bots don't have the required config.
+
+// TODO(leozwang): Because ExternalMediaProcessing is disabled in webrtc,
+// disable this unit test on Android for now.
+#if !defined(OS_ANDROID)

[tommi (sloooow) - chröme, 2013/02/11 09:52:58]

Instead of this, can you use the MAYBE_ approach as we do elsewhere?  This means
that MAYBE_StartRecording will be defined as StartRecording on all platforms
except Android.  On Android it will be defined as DISABLED_StartRecording.

The test will continue to be compiled for all platforms and you can still run it
manually if you want by overriding the disabled check.

[leozwang1, 2013/02/11 18:03:43]

Done.

 TEST_F(WebRTCAudioDeviceTest, StartRecording) {
   if (!has_input_devices_ || !has_output_devices_) {
     LOG(WARNING) << "Missing audio devices.";
@@ -388,6 +392,7 @@
   EXPECT_EQ(0, base->DeleteChannel(ch));
   EXPECT_EQ(0, base->Terminate());
 }
+#endif
 
 // Uses WebRtcAudioDeviceImpl to play a local wave file.
 // Disabled when running headless since the bots don't have the required config.
@@ -505,11 +510,19 @@
 
   ScopedWebRTCPtr<webrtc::VoEAudioProcessing> audio_processing(engine.get());
   ASSERT_TRUE(audio_processing.valid());
+#if defined(OS_ANDROID)
+  // On Android, by default AGC is off.
+  bool enabled = true;
+  webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
+  EXPECT_EQ(0, audio_processing->GetAgcStatus(enabled, agc_mode));
+  EXPECT_FALSE(enabled);
+#else
   bool enabled = false;
-  webrtc::AgcModes agc_mode =  webrtc::kAgcDefault;
+  webrtc::AgcModes agc_mode = webrtc::kAgcDefault;
   EXPECT_EQ(0, audio_processing->GetAgcStatus(enabled, agc_mode));
   EXPECT_TRUE(enabled);
   EXPECT_EQ(agc_mode, webrtc::kAgcAdaptiveAnalog);
+#endif
 
   int ch = base->CreateChannel();
   EXPECT_NE(-1, ch);