Chromium Code Reviews| 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); |
| } |