| 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 1ae465cbeff8aadc3edef583bc511646364a0e53..fedf5ee49bb7d5f3943f2cc2db1f24560ae0f289 100644
|
| --- a/webrtc/modules/audio_processing/test/audio_processing_simulator.h
|
| +++ b/webrtc/modules/audio_processing/test/audio_processing_simulator.h
|
| @@ -19,6 +19,7 @@
|
|
|
| #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"
|
| #include "webrtc/rtc_base/constructormagic.h"
|
| #include "webrtc/rtc_base/optional.h"
|
| @@ -76,6 +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;
|
| @@ -166,6 +170,7 @@ class AudioProcessingSimulator {
|
| AudioFrame rev_frame_;
|
| AudioFrame fwd_frame_;
|
| bool bitexact_output_ = true;
|
| + rtc::Optional<int> aec_dump_mic_level_;
|
|
|
| private:
|
| void SetupOutput();
|
| @@ -178,6 +183,9 @@ class AudioProcessingSimulator {
|
| TickIntervalStats proc_time_;
|
| std::ofstream residual_echo_likelihood_graph_writer_;
|
|
|
| + int analog_mic_level_;
|
| + FakeRecordingDevice fake_recording_device_;
|
| +
|
| rtc::TaskQueue worker_queue_;
|
|
|
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator);
|
|
|