Index: webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc |
diff --git a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc |
index 3d2c71fb2dbef38993eef42054aa2b27db9e6f23..52c5643d2db670fab5bf7516addc5efa4ab23a88 100644 |
--- a/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc |
+++ b/webrtc/modules/audio_processing/audio_processing_impl_locking_unittest.cc |
@@ -596,7 +596,6 @@ bool StatsProcessor::Process() { |
(test_config_->aec_type == |
AecType::BasicWebRtcAecSettingsWithAecMobile)); |
EXPECT_TRUE(apm_->gain_control()->is_enabled()); |
- apm_->gain_control()->stream_analog_level(); |
EXPECT_TRUE(apm_->noise_suppression()->is_enabled()); |
// The below return values are not testable. |
@@ -709,9 +708,12 @@ void CaptureProcessor::CallApmCaptureSide() { |
// Prepare a proper capture side processing API call input. |
PrepareFrame(); |
- // Set the stream delay |
+ // Set the stream delay. |
apm_->set_stream_delay_ms(30); |
+ // Set the analog level. |
+ apm_->gain_control()->set_stream_analog_level(80); |
+ |
// Call the specified capture side API processing method. |
int result = AudioProcessing::kNoError; |
switch (test_config_->capture_api_function) { |
@@ -734,6 +736,9 @@ void CaptureProcessor::CallApmCaptureSide() { |
FAIL(); |
} |
+ // Retrieve the new analog level. |
+ apm_->gain_control()->stream_analog_level(); |
the sun
2016/02/09 11:49:55
Just for testing? It's a bit confusing that you sa
peah-webrtc
2016/02/10 09:33:37
The problem with that is that the purpose is not a
|
+ |
// Check the return code for error. |
ASSERT_EQ(AudioProcessing::kNoError, result); |
} |