Index: webrtc/modules/audio_processing/test/aec_dump_based_simulator.h |
diff --git a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h |
index 5f7d138aa41eff5b2bf5a3f9374ba99836939fb5..188850bf7278934af4a5672fb5d25fac62cde805 100644 |
--- a/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h |
+++ b/webrtc/modules/audio_processing/test/aec_dump_based_simulator.h |
@@ -11,6 +11,8 @@ |
#ifndef WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ |
#define WEBRTC_MODULES_AUDIO_PROCESSING_TEST_AEC_DUMP_BASED_SIMULATOR_H_ |
+#include <memory> |
+ |
#include "webrtc/modules/audio_processing/test/audio_processing_simulator.h" |
#include "webrtc/base/constructormagic.h" |
@@ -41,8 +43,7 @@ class AecDumpBasedSimulator final : public AudioProcessingSimulator { |
void HandleMessage(const webrtc::audioproc::Stream& msg); |
void HandleMessage(const webrtc::audioproc::ReverseStream& msg); |
void HandleMessage(const webrtc::audioproc::Config& msg); |
- void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg, |
- bool* set_stream_analog_level_called); |
+ void PrepareProcessStreamCall(const webrtc::audioproc::Stream& msg); |
void PrepareReverseProcessStreamCall( |
const webrtc::audioproc::ReverseStream& msg); |
void VerifyProcessStreamBitExactness(const webrtc::audioproc::Stream& msg); |
@@ -59,6 +60,10 @@ class AecDumpBasedSimulator final : public AudioProcessingSimulator { |
bool artificial_nearend_eof_reported_ = false; |
InterfaceType interface_used_ = InterfaceType::kNotSpecified; |
+ // TODO(aleloi): Remove once a FakeRecordingDevice is addded as protected |
+ // member of AudioProcessingSimulator. |
+ int last_specified_microphone_level_ = kInitialMicrophoneGainLevel; |
+ |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AecDumpBasedSimulator); |
}; |