Index: webrtc/modules/audio_processing/test/wav_based_simulator.cc |
diff --git a/webrtc/modules/audio_processing/test/wav_based_simulator.cc b/webrtc/modules/audio_processing/test/wav_based_simulator.cc |
index 6cf0b744e6f5c040c81f7b8798e0992008c96b63..f5ad4b280e70349bf3ae7e1455c6bdfb1f42a48f 100644 |
--- a/webrtc/modules/audio_processing/test/wav_based_simulator.cc |
+++ b/webrtc/modules/audio_processing/test/wav_based_simulator.cc |
@@ -12,6 +12,8 @@ |
#include <stdio.h> |
#include <iostream> |
+#include <memory> |
+#include <utility> |
#include "webrtc/base/checks.h" |
#include "webrtc/modules/audio_processing/test/test_utils.h" |
@@ -79,10 +81,6 @@ void WavBasedSimulator::PrepareProcessStreamCall() { |
ap_->echo_cancellation()->set_stream_drift_samples( |
settings_.stream_drift_samples ? *settings_.stream_drift_samples : 0); |
peah-webrtc
2017/04/26 12:54:44
What you do here is that you move the set_stream_a
|
- |
- RTC_CHECK_EQ(AudioProcessing::kNoError, |
- ap_->gain_control()->set_stream_analog_level( |
- last_specified_microphone_level_)); |
} |
void WavBasedSimulator::PrepareReverseProcessStreamCall() { |
@@ -143,10 +141,6 @@ bool WavBasedSimulator::HandleProcessStreamCall() { |
if (samples_left_to_process) { |
PrepareProcessStreamCall(); |
ProcessStream(settings_.fixed_interface); |
- // Call stream analog level to ensure that any side-effects are triggered. |
- (void)ap_->gain_control()->stream_analog_level(); |
- last_specified_microphone_level_ = |
- ap_->gain_control()->stream_analog_level(); |
} |
return samples_left_to_process; |
} |