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 c9ac2e377bb14c631cb2f31b45501f4b3c2c5eae..6cbb6b3f201eae274afd81dee26e7a03b997ae9b 100644 |
--- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
+++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h |
@@ -17,9 +17,9 @@ |
#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/test_utils.h" |
@@ -27,6 +27,8 @@ |
namespace webrtc { |
namespace test { |
+class FakeRecordingDevice; |
+ |
// Holds all the parameters available for controlling the simulation. |
struct SimulationSettings { |
SimulationSettings(); |
@@ -75,6 +77,9 @@ struct SimulationSettings { |
rtc::Optional<int> vad_likelihood; |
rtc::Optional<int> ns_level; |
rtc::Optional<bool> use_refined_adaptive_filter; |
+ int initial_mic_level; |
+ bool simulate_mic_gain = false; |
+ rtc::Optional<int> simulated_mic_kind; |
bool report_performance = false; |
bool report_bitexactness = false; |
bool use_verbose_logging = false; |
@@ -165,6 +170,7 @@ class AudioProcessingSimulator { |
AudioFrame rev_frame_; |
AudioFrame fwd_frame_; |
bool bitexact_output_ = true; |
+ rtc::Optional<int> aec_dump_mic_level_; |
private: |
void SetupOutput(); |
@@ -176,6 +182,7 @@ class AudioProcessingSimulator { |
std::unique_ptr<ChannelBufferWavWriter> reverse_buffer_writer_; |
TickIntervalStats proc_time_; |
std::ofstream residual_echo_likelihood_graph_writer_; |
+ std::unique_ptr<FakeRecordingDevice> fake_recording_device_; |
RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator); |
}; |