Chromium Code Reviews| 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_; |
|
peah-webrtc
2017/09/15 09:36:20
Please remove the empty line on 185
AleBzk
2017/09/22 12:33:55
Done.
|
| + int analog_mic_level_; |
| + FakeRecordingDevice fake_recording_device_; |
| + |
| rtc::TaskQueue worker_queue_; |
| RTC_DISALLOW_IMPLICIT_CONSTRUCTORS(AudioProcessingSimulator); |