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 +322,14 @@ |
// 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) |
+#define MAYBE_StartRecording DISABLED_StartRecording |
+#else |
+#define MAYBE_StartRecording StartRecording |
+#endif |
TEST_F(WebRTCAudioDeviceTest, StartRecording) { |
if (!has_input_devices_ || !has_output_devices_) { |
LOG(WARNING) << "Missing audio devices."; |
@@ -505,11 +513,19 @@ |
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); |