Index: webrtc/modules/audio_processing/test/audio_processing_simulator.h |
diff --git a/webrtc/modules/audio_processing/test/audio_processing_simulator.h b/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
index db3c92f942a7bdda6d2903f40d164c69f33bd7da..dacc88383add35fa56b912222b1e724b187a43ad 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
@@ -17,11 +17,12 @@ |
#include <memory> |
#include <string> |
-#include "webrtc/base/timeutils.h" |
#include "webrtc/base/constructormagic.h" |
#include "webrtc/base/optional.h" |
+#include "webrtc/base/timeutils.h" |
#include "webrtc/common_audio/channel_buffer.h" |
#include "webrtc/modules/audio_processing/include/audio_processing.h" |
+#include "webrtc/modules/audio_processing/test/fake_recording_device.h" |
#include "webrtc/modules/audio_processing/test/test_utils.h" |
namespace webrtc { |
@@ -79,6 +80,7 @@ struct SimulationSettings { |
rtc::Optional<int> ns_level; |
rtc::Optional<bool> use_refined_adaptive_filter; |
bool simulate_mic_gain = false; |
+ rtc::Optional<int> simulated_mic_kind; |
bool report_performance = false; |
bool report_bitexactness = false; |
bool use_verbose_logging = false; |
@@ -169,6 +171,7 @@ class AudioProcessingSimulator { |
AudioFrame rev_frame_; |
AudioFrame fwd_frame_; |
bool bitexact_output_ = true; |
+ rtc::Optional<FakeRecordingDevice> fake_recording_device_; |
private: |
void SetupOutput(); |